* [Qemu-devel] [PATCH 01/12] audio: remove esd backend
2015-06-01 13:23 [Qemu-devel] [PATCH 00/12] Audio backend cleanup Kővágó, Zoltán
@ 2015-06-01 13:23 ` Kővágó, Zoltán
2015-06-01 13:23 ` [Qemu-devel] [PATCH 02/12] audio: remove fmod backend Kővágó, Zoltán
` (11 subsequent siblings)
12 siblings, 0 replies; 26+ messages in thread
From: Kővágó, Zoltán @ 2015-06-01 13:23 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
ESD is no longer developed and replaced by PulseAudio.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/Makefile.objs | 1 -
audio/audio_int.h | 1 -
audio/esdaudio.c | 557 ----------------------------------------------------
configure | 18 +-
4 files changed, 6 insertions(+), 571 deletions(-)
delete mode 100644 audio/esdaudio.c
diff --git a/audio/Makefile.objs b/audio/Makefile.objs
index 26a0ac9..5573ac1 100644
--- a/audio/Makefile.objs
+++ b/audio/Makefile.objs
@@ -6,7 +6,6 @@ common-obj-$(CONFIG_COREAUDIO) += coreaudio.o
common-obj-$(CONFIG_ALSA) += alsaaudio.o
common-obj-$(CONFIG_DSOUND) += dsoundaudio.o
common-obj-$(CONFIG_FMOD) += fmodaudio.o
-common-obj-$(CONFIG_ESD) += esdaudio.o
common-obj-$(CONFIG_PA) += paaudio.o
common-obj-$(CONFIG_WINWAVE) += winwaveaudio.o
common-obj-$(CONFIG_AUDIO_PT_INT) += audio_pt_int.o
diff --git a/audio/audio_int.h b/audio/audio_int.h
index fd019a0..9dd6b7f 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -210,7 +210,6 @@ extern struct audio_driver fmod_audio_driver;
extern struct audio_driver alsa_audio_driver;
extern struct audio_driver coreaudio_audio_driver;
extern struct audio_driver dsound_audio_driver;
-extern struct audio_driver esd_audio_driver;
extern struct audio_driver pa_audio_driver;
extern struct audio_driver spice_audio_driver;
extern struct audio_driver winwave_audio_driver;
diff --git a/audio/esdaudio.c b/audio/esdaudio.c
deleted file mode 100644
index eea9cce..0000000
--- a/audio/esdaudio.c
+++ /dev/null
@@ -1,557 +0,0 @@
-/*
- * QEMU ESD audio driver
- *
- * Copyright (c) 2006 Frederick Reeve (brushed up by malc)
- *
- * Permission is hereby granted, free of charge, to any person obtaining a copy
- * of this software and associated documentation files (the "Software"), to deal
- * in the Software without restriction, including without limitation the rights
- * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
- * copies of the Software, and to permit persons to whom the Software is
- * furnished to do so, subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be included in
- * all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
- * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
- * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
- * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
- * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
- * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
- * THE SOFTWARE.
- */
-#include <esd.h>
-#include "qemu-common.h"
-#include "audio.h"
-
-#define AUDIO_CAP "esd"
-#include "audio_int.h"
-#include "audio_pt_int.h"
-
-typedef struct {
- HWVoiceOut hw;
- int done;
- int live;
- int decr;
- int rpos;
- void *pcm_buf;
- int fd;
- struct audio_pt pt;
-} ESDVoiceOut;
-
-typedef struct {
- HWVoiceIn hw;
- int done;
- int dead;
- int incr;
- int wpos;
- void *pcm_buf;
- int fd;
- struct audio_pt pt;
-} ESDVoiceIn;
-
-static struct {
- int samples;
- int divisor;
- char *dac_host;
- char *adc_host;
-} conf = {
- .samples = 1024,
- .divisor = 2,
-};
-
-static void GCC_FMT_ATTR (2, 3) qesd_logerr (int err, const char *fmt, ...)
-{
- va_list ap;
-
- va_start (ap, fmt);
- AUD_vlog (AUDIO_CAP, fmt, ap);
- va_end (ap);
-
- AUD_log (AUDIO_CAP, "Reason: %s\n", strerror (err));
-}
-
-/* playback */
-static void *qesd_thread_out (void *arg)
-{
- ESDVoiceOut *esd = arg;
- HWVoiceOut *hw = &esd->hw;
- int threshold;
-
- threshold = conf.divisor ? hw->samples / conf.divisor : 0;
-
- if (audio_pt_lock (&esd->pt, AUDIO_FUNC)) {
- return NULL;
- }
-
- for (;;) {
- int decr, to_mix, rpos;
-
- for (;;) {
- if (esd->done) {
- goto exit;
- }
-
- if (esd->live > threshold) {
- break;
- }
-
- if (audio_pt_wait (&esd->pt, AUDIO_FUNC)) {
- goto exit;
- }
- }
-
- decr = to_mix = esd->live;
- rpos = hw->rpos;
-
- if (audio_pt_unlock (&esd->pt, AUDIO_FUNC)) {
- return NULL;
- }
-
- while (to_mix) {
- ssize_t written;
- int chunk = audio_MIN (to_mix, hw->samples - rpos);
- struct st_sample *src = hw->mix_buf + rpos;
-
- hw->clip (esd->pcm_buf, src, chunk);
-
- again:
- written = write (esd->fd, esd->pcm_buf, chunk << hw->info.shift);
- if (written == -1) {
- if (errno == EINTR || errno == EAGAIN) {
- goto again;
- }
- qesd_logerr (errno, "write failed\n");
- return NULL;
- }
-
- if (written != chunk << hw->info.shift) {
- int wsamples = written >> hw->info.shift;
- int wbytes = wsamples << hw->info.shift;
- if (wbytes != written) {
- dolog ("warning: Misaligned write %d (requested %zd), "
- "alignment %d\n",
- wbytes, written, hw->info.align + 1);
- }
- to_mix -= wsamples;
- rpos = (rpos + wsamples) % hw->samples;
- break;
- }
-
- rpos = (rpos + chunk) % hw->samples;
- to_mix -= chunk;
- }
-
- if (audio_pt_lock (&esd->pt, AUDIO_FUNC)) {
- return NULL;
- }
-
- esd->rpos = rpos;
- esd->live -= decr;
- esd->decr += decr;
- }
-
- exit:
- audio_pt_unlock (&esd->pt, AUDIO_FUNC);
- return NULL;
-}
-
-static int qesd_run_out (HWVoiceOut *hw, int live)
-{
- int decr;
- ESDVoiceOut *esd = (ESDVoiceOut *) hw;
-
- if (audio_pt_lock (&esd->pt, AUDIO_FUNC)) {
- return 0;
- }
-
- decr = audio_MIN (live, esd->decr);
- esd->decr -= decr;
- esd->live = live - decr;
- hw->rpos = esd->rpos;
- if (esd->live > 0) {
- audio_pt_unlock_and_signal (&esd->pt, AUDIO_FUNC);
- }
- else {
- audio_pt_unlock (&esd->pt, AUDIO_FUNC);
- }
- return decr;
-}
-
-static int qesd_write (SWVoiceOut *sw, void *buf, int len)
-{
- return audio_pcm_sw_write (sw, buf, len);
-}
-
-static int qesd_init_out (HWVoiceOut *hw, struct audsettings *as)
-{
- ESDVoiceOut *esd = (ESDVoiceOut *) hw;
- struct audsettings obt_as = *as;
- int esdfmt = ESD_STREAM | ESD_PLAY;
-
- esdfmt |= (as->nchannels == 2) ? ESD_STEREO : ESD_MONO;
- switch (as->fmt) {
- case AUD_FMT_S8:
- case AUD_FMT_U8:
- esdfmt |= ESD_BITS8;
- obt_as.fmt = AUD_FMT_U8;
- break;
-
- case AUD_FMT_S32:
- case AUD_FMT_U32:
- dolog ("Will use 16 instead of 32 bit samples\n");
- /* fall through */
- case AUD_FMT_S16:
- case AUD_FMT_U16:
- deffmt:
- esdfmt |= ESD_BITS16;
- obt_as.fmt = AUD_FMT_S16;
- break;
-
- default:
- dolog ("Internal logic error: Bad audio format %d\n", as->fmt);
- goto deffmt;
-
- }
- obt_as.endianness = AUDIO_HOST_ENDIANNESS;
-
- audio_pcm_init_info (&hw->info, &obt_as);
-
- hw->samples = conf.samples;
- esd->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
- if (!esd->pcm_buf) {
- dolog ("Could not allocate buffer (%d bytes)\n",
- hw->samples << hw->info.shift);
- return -1;
- }
-
- esd->fd = esd_play_stream (esdfmt, as->freq, conf.dac_host, NULL);
- if (esd->fd < 0) {
- qesd_logerr (errno, "esd_play_stream failed\n");
- goto fail1;
- }
-
- if (audio_pt_init (&esd->pt, qesd_thread_out, esd, AUDIO_CAP, AUDIO_FUNC)) {
- goto fail2;
- }
-
- return 0;
-
- fail2:
- if (close (esd->fd)) {
- qesd_logerr (errno, "%s: close on esd socket(%d) failed\n",
- AUDIO_FUNC, esd->fd);
- }
- esd->fd = -1;
-
- fail1:
- g_free (esd->pcm_buf);
- esd->pcm_buf = NULL;
- return -1;
-}
-
-static void qesd_fini_out (HWVoiceOut *hw)
-{
- void *ret;
- ESDVoiceOut *esd = (ESDVoiceOut *) hw;
-
- audio_pt_lock (&esd->pt, AUDIO_FUNC);
- esd->done = 1;
- audio_pt_unlock_and_signal (&esd->pt, AUDIO_FUNC);
- audio_pt_join (&esd->pt, &ret, AUDIO_FUNC);
-
- if (esd->fd >= 0) {
- if (close (esd->fd)) {
- qesd_logerr (errno, "failed to close esd socket\n");
- }
- esd->fd = -1;
- }
-
- audio_pt_fini (&esd->pt, AUDIO_FUNC);
-
- g_free (esd->pcm_buf);
- esd->pcm_buf = NULL;
-}
-
-static int qesd_ctl_out (HWVoiceOut *hw, int cmd, ...)
-{
- (void) hw;
- (void) cmd;
- return 0;
-}
-
-/* capture */
-static void *qesd_thread_in (void *arg)
-{
- ESDVoiceIn *esd = arg;
- HWVoiceIn *hw = &esd->hw;
- int threshold;
-
- threshold = conf.divisor ? hw->samples / conf.divisor : 0;
-
- if (audio_pt_lock (&esd->pt, AUDIO_FUNC)) {
- return NULL;
- }
-
- for (;;) {
- int incr, to_grab, wpos;
-
- for (;;) {
- if (esd->done) {
- goto exit;
- }
-
- if (esd->dead > threshold) {
- break;
- }
-
- if (audio_pt_wait (&esd->pt, AUDIO_FUNC)) {
- goto exit;
- }
- }
-
- incr = to_grab = esd->dead;
- wpos = hw->wpos;
-
- if (audio_pt_unlock (&esd->pt, AUDIO_FUNC)) {
- return NULL;
- }
-
- while (to_grab) {
- ssize_t nread;
- int chunk = audio_MIN (to_grab, hw->samples - wpos);
- void *buf = advance (esd->pcm_buf, wpos);
-
- again:
- nread = read (esd->fd, buf, chunk << hw->info.shift);
- if (nread == -1) {
- if (errno == EINTR || errno == EAGAIN) {
- goto again;
- }
- qesd_logerr (errno, "read failed\n");
- return NULL;
- }
-
- if (nread != chunk << hw->info.shift) {
- int rsamples = nread >> hw->info.shift;
- int rbytes = rsamples << hw->info.shift;
- if (rbytes != nread) {
- dolog ("warning: Misaligned write %d (requested %zd), "
- "alignment %d\n",
- rbytes, nread, hw->info.align + 1);
- }
- to_grab -= rsamples;
- wpos = (wpos + rsamples) % hw->samples;
- break;
- }
-
- hw->conv (hw->conv_buf + wpos, buf, nread >> hw->info.shift);
- wpos = (wpos + chunk) % hw->samples;
- to_grab -= chunk;
- }
-
- if (audio_pt_lock (&esd->pt, AUDIO_FUNC)) {
- return NULL;
- }
-
- esd->wpos = wpos;
- esd->dead -= incr;
- esd->incr += incr;
- }
-
- exit:
- audio_pt_unlock (&esd->pt, AUDIO_FUNC);
- return NULL;
-}
-
-static int qesd_run_in (HWVoiceIn *hw)
-{
- int live, incr, dead;
- ESDVoiceIn *esd = (ESDVoiceIn *) hw;
-
- if (audio_pt_lock (&esd->pt, AUDIO_FUNC)) {
- return 0;
- }
-
- live = audio_pcm_hw_get_live_in (hw);
- dead = hw->samples - live;
- incr = audio_MIN (dead, esd->incr);
- esd->incr -= incr;
- esd->dead = dead - incr;
- hw->wpos = esd->wpos;
- if (esd->dead > 0) {
- audio_pt_unlock_and_signal (&esd->pt, AUDIO_FUNC);
- }
- else {
- audio_pt_unlock (&esd->pt, AUDIO_FUNC);
- }
- return incr;
-}
-
-static int qesd_read (SWVoiceIn *sw, void *buf, int len)
-{
- return audio_pcm_sw_read (sw, buf, len);
-}
-
-static int qesd_init_in (HWVoiceIn *hw, struct audsettings *as)
-{
- ESDVoiceIn *esd = (ESDVoiceIn *) hw;
- struct audsettings obt_as = *as;
- int esdfmt = ESD_STREAM | ESD_RECORD;
-
- esdfmt |= (as->nchannels == 2) ? ESD_STEREO : ESD_MONO;
- switch (as->fmt) {
- case AUD_FMT_S8:
- case AUD_FMT_U8:
- esdfmt |= ESD_BITS8;
- obt_as.fmt = AUD_FMT_U8;
- break;
-
- case AUD_FMT_S16:
- case AUD_FMT_U16:
- esdfmt |= ESD_BITS16;
- obt_as.fmt = AUD_FMT_S16;
- break;
-
- case AUD_FMT_S32:
- case AUD_FMT_U32:
- dolog ("Will use 16 instead of 32 bit samples\n");
- esdfmt |= ESD_BITS16;
- obt_as.fmt = AUD_FMT_S16;
- break;
- }
- obt_as.endianness = AUDIO_HOST_ENDIANNESS;
-
- audio_pcm_init_info (&hw->info, &obt_as);
-
- hw->samples = conf.samples;
- esd->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
- if (!esd->pcm_buf) {
- dolog ("Could not allocate buffer (%d bytes)\n",
- hw->samples << hw->info.shift);
- return -1;
- }
-
- esd->fd = esd_record_stream (esdfmt, as->freq, conf.adc_host, NULL);
- if (esd->fd < 0) {
- qesd_logerr (errno, "esd_record_stream failed\n");
- goto fail1;
- }
-
- if (audio_pt_init (&esd->pt, qesd_thread_in, esd, AUDIO_CAP, AUDIO_FUNC)) {
- goto fail2;
- }
-
- return 0;
-
- fail2:
- if (close (esd->fd)) {
- qesd_logerr (errno, "%s: close on esd socket(%d) failed\n",
- AUDIO_FUNC, esd->fd);
- }
- esd->fd = -1;
-
- fail1:
- g_free (esd->pcm_buf);
- esd->pcm_buf = NULL;
- return -1;
-}
-
-static void qesd_fini_in (HWVoiceIn *hw)
-{
- void *ret;
- ESDVoiceIn *esd = (ESDVoiceIn *) hw;
-
- audio_pt_lock (&esd->pt, AUDIO_FUNC);
- esd->done = 1;
- audio_pt_unlock_and_signal (&esd->pt, AUDIO_FUNC);
- audio_pt_join (&esd->pt, &ret, AUDIO_FUNC);
-
- if (esd->fd >= 0) {
- if (close (esd->fd)) {
- qesd_logerr (errno, "failed to close esd socket\n");
- }
- esd->fd = -1;
- }
-
- audio_pt_fini (&esd->pt, AUDIO_FUNC);
-
- g_free (esd->pcm_buf);
- esd->pcm_buf = NULL;
-}
-
-static int qesd_ctl_in (HWVoiceIn *hw, int cmd, ...)
-{
- (void) hw;
- (void) cmd;
- return 0;
-}
-
-/* common */
-static void *qesd_audio_init (void)
-{
- return &conf;
-}
-
-static void qesd_audio_fini (void *opaque)
-{
- (void) opaque;
- ldebug ("esd_fini");
-}
-
-struct audio_option qesd_options[] = {
- {
- .name = "SAMPLES",
- .tag = AUD_OPT_INT,
- .valp = &conf.samples,
- .descr = "buffer size in samples"
- },
- {
- .name = "DIVISOR",
- .tag = AUD_OPT_INT,
- .valp = &conf.divisor,
- .descr = "threshold divisor"
- },
- {
- .name = "DAC_HOST",
- .tag = AUD_OPT_STR,
- .valp = &conf.dac_host,
- .descr = "playback host"
- },
- {
- .name = "ADC_HOST",
- .tag = AUD_OPT_STR,
- .valp = &conf.adc_host,
- .descr = "capture host"
- },
- { /* End of list */ }
-};
-
-static struct audio_pcm_ops qesd_pcm_ops = {
- .init_out = qesd_init_out,
- .fini_out = qesd_fini_out,
- .run_out = qesd_run_out,
- .write = qesd_write,
- .ctl_out = qesd_ctl_out,
-
- .init_in = qesd_init_in,
- .fini_in = qesd_fini_in,
- .run_in = qesd_run_in,
- .read = qesd_read,
- .ctl_in = qesd_ctl_in,
-};
-
-struct audio_driver esd_audio_driver = {
- .name = "esd",
- .descr = "http://en.wikipedia.org/wiki/Esound",
- .options = qesd_options,
- .init = qesd_audio_init,
- .fini = qesd_audio_fini,
- .pcm_ops = &qesd_pcm_ops,
- .can_be_default = 0,
- .max_voices_out = INT_MAX,
- .max_voices_in = INT_MAX,
- .voice_size_out = sizeof (ESDVoiceOut),
- .voice_size_in = sizeof (ESDVoiceIn)
-};
diff --git a/configure b/configure
index 4e2f78a..acd471d 100755
--- a/configure
+++ b/configure
@@ -575,13 +575,13 @@ MINGW32*)
GNU/kFreeBSD)
bsd="yes"
audio_drv_list="oss"
- audio_possible_drivers="oss sdl esd pa"
+ audio_possible_drivers="oss sdl pa"
;;
FreeBSD)
bsd="yes"
make="${MAKE-gmake}"
audio_drv_list="oss"
- audio_possible_drivers="oss sdl esd pa"
+ audio_possible_drivers="oss sdl pa"
# needed for kinfo_getvmmap(3) in libutil.h
LIBS="-lutil $LIBS"
netmap="" # enable netmap autodetect
@@ -591,14 +591,14 @@ DragonFly)
bsd="yes"
make="${MAKE-gmake}"
audio_drv_list="oss"
- audio_possible_drivers="oss sdl esd pa"
+ audio_possible_drivers="oss sdl pa"
HOST_VARIANT_DIR="dragonfly"
;;
NetBSD)
bsd="yes"
make="${MAKE-gmake}"
audio_drv_list="oss"
- audio_possible_drivers="oss sdl esd"
+ audio_possible_drivers="oss sdl"
oss_lib="-lossaudio"
HOST_VARIANT_DIR="netbsd"
;;
@@ -606,7 +606,7 @@ OpenBSD)
bsd="yes"
make="${MAKE-gmake}"
audio_drv_list="sdl"
- audio_possible_drivers="sdl esd"
+ audio_possible_drivers="sdl"
HOST_VARIANT_DIR="openbsd"
;;
Darwin)
@@ -674,7 +674,7 @@ Haiku)
;;
*)
audio_drv_list="oss"
- audio_possible_drivers="oss alsa sdl esd pa"
+ audio_possible_drivers="oss alsa sdl pa"
linux="yes"
linux_user="yes"
kvm="yes"
@@ -2630,12 +2630,6 @@ for drv in $audio_drv_list; do
libs_softmmu="$fmod_lib $libs_softmmu"
;;
- esd)
- audio_drv_probe $drv esd.h -lesd 'return esd_play_stream(0, 0, "", 0);'
- libs_softmmu="-lesd $libs_softmmu"
- audio_pt_int="yes"
- ;;
-
pa)
audio_drv_probe $drv pulse/mainloop.h "-lpulse" \
"pa_mainloop *m = 0; pa_mainloop_free (m); return 0;"
--
2.4.2
^ permalink raw reply related [flat|nested] 26+ messages in thread* [Qemu-devel] [PATCH 02/12] audio: remove fmod backend
2015-06-01 13:23 [Qemu-devel] [PATCH 00/12] Audio backend cleanup Kővágó, Zoltán
2015-06-01 13:23 ` [Qemu-devel] [PATCH 01/12] audio: remove esd backend Kővágó, Zoltán
@ 2015-06-01 13:23 ` Kővágó, Zoltán
2015-06-01 13:23 ` [Qemu-devel] [PATCH 03/12] audio: remove sdl backend Kővágó, Zoltán
` (10 subsequent siblings)
12 siblings, 0 replies; 26+ messages in thread
From: Kővágó, Zoltán @ 2015-06-01 13:23 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/Makefile.objs | 2 -
audio/audio_int.h | 1 -
audio/fmodaudio.c | 685 ----------------------------------------------------
configure | 27 +--
4 files changed, 2 insertions(+), 713 deletions(-)
delete mode 100644 audio/fmodaudio.c
diff --git a/audio/Makefile.objs b/audio/Makefile.objs
index 5573ac1..b4c0608 100644
--- a/audio/Makefile.objs
+++ b/audio/Makefile.objs
@@ -5,12 +5,10 @@ common-obj-$(CONFIG_SPICE) += spiceaudio.o
common-obj-$(CONFIG_COREAUDIO) += coreaudio.o
common-obj-$(CONFIG_ALSA) += alsaaudio.o
common-obj-$(CONFIG_DSOUND) += dsoundaudio.o
-common-obj-$(CONFIG_FMOD) += fmodaudio.o
common-obj-$(CONFIG_PA) += paaudio.o
common-obj-$(CONFIG_WINWAVE) += winwaveaudio.o
common-obj-$(CONFIG_AUDIO_PT_INT) += audio_pt_int.o
common-obj-$(CONFIG_AUDIO_WIN_INT) += audio_win_int.o
common-obj-y += wavcapture.o
-$(obj)/audio.o $(obj)/fmodaudio.o: QEMU_CFLAGS += $(FMOD_CFLAGS)
sdlaudio.o-cflags := $(SDL_CFLAGS)
diff --git a/audio/audio_int.h b/audio/audio_int.h
index 9dd6b7f..7445602 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -206,7 +206,6 @@ extern struct audio_driver no_audio_driver;
extern struct audio_driver oss_audio_driver;
extern struct audio_driver sdl_audio_driver;
extern struct audio_driver wav_audio_driver;
-extern struct audio_driver fmod_audio_driver;
extern struct audio_driver alsa_audio_driver;
extern struct audio_driver coreaudio_audio_driver;
extern struct audio_driver dsound_audio_driver;
diff --git a/audio/fmodaudio.c b/audio/fmodaudio.c
deleted file mode 100644
index fabf84d..0000000
--- a/audio/fmodaudio.c
+++ /dev/null
@@ -1,685 +0,0 @@
-/*
- * QEMU FMOD audio driver
- *
- * Copyright (c) 2004-2005 Vassili Karpov (malc)
- *
- * Permission is hereby granted, free of charge, to any person obtaining a copy
- * of this software and associated documentation files (the "Software"), to deal
- * in the Software without restriction, including without limitation the rights
- * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
- * copies of the Software, and to permit persons to whom the Software is
- * furnished to do so, subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be included in
- * all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
- * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
- * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
- * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
- * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
- * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
- * THE SOFTWARE.
- */
-#include <fmod.h>
-#include <fmod_errors.h>
-#include "qemu-common.h"
-#include "audio.h"
-
-#define AUDIO_CAP "fmod"
-#include "audio_int.h"
-
-typedef struct FMODVoiceOut {
- HWVoiceOut hw;
- unsigned int old_pos;
- FSOUND_SAMPLE *fmod_sample;
- int channel;
-} FMODVoiceOut;
-
-typedef struct FMODVoiceIn {
- HWVoiceIn hw;
- FSOUND_SAMPLE *fmod_sample;
-} FMODVoiceIn;
-
-static struct {
- const char *drvname;
- int nb_samples;
- int freq;
- int nb_channels;
- int bufsize;
- int broken_adc;
-} conf = {
- .nb_samples = 2048 * 2,
- .freq = 44100,
- .nb_channels = 2,
-};
-
-static void GCC_FMT_ATTR (1, 2) fmod_logerr (const char *fmt, ...)
-{
- va_list ap;
-
- va_start (ap, fmt);
- AUD_vlog (AUDIO_CAP, fmt, ap);
- va_end (ap);
-
- AUD_log (AUDIO_CAP, "Reason: %s\n",
- FMOD_ErrorString (FSOUND_GetError ()));
-}
-
-static void GCC_FMT_ATTR (2, 3) fmod_logerr2 (
- const char *typ,
- const char *fmt,
- ...
- )
-{
- va_list ap;
-
- AUD_log (AUDIO_CAP, "Could not initialize %s\n", typ);
-
- va_start (ap, fmt);
- AUD_vlog (AUDIO_CAP, fmt, ap);
- va_end (ap);
-
- AUD_log (AUDIO_CAP, "Reason: %s\n",
- FMOD_ErrorString (FSOUND_GetError ()));
-}
-
-static int fmod_write (SWVoiceOut *sw, void *buf, int len)
-{
- return audio_pcm_sw_write (sw, buf, len);
-}
-
-static void fmod_clear_sample (FMODVoiceOut *fmd)
-{
- HWVoiceOut *hw = &fmd->hw;
- int status;
- void *p1 = 0, *p2 = 0;
- unsigned int len1 = 0, len2 = 0;
-
- status = FSOUND_Sample_Lock (
- fmd->fmod_sample,
- 0,
- hw->samples << hw->info.shift,
- &p1,
- &p2,
- &len1,
- &len2
- );
-
- if (!status) {
- fmod_logerr ("Failed to lock sample\n");
- return;
- }
-
- if ((len1 & hw->info.align) || (len2 & hw->info.align)) {
- dolog ("Lock returned misaligned length %d, %d, alignment %d\n",
- len1, len2, hw->info.align + 1);
- goto fail;
- }
-
- if ((len1 + len2) - (hw->samples << hw->info.shift)) {
- dolog ("Lock returned incomplete length %d, %d\n",
- len1 + len2, hw->samples << hw->info.shift);
- goto fail;
- }
-
- audio_pcm_info_clear_buf (&hw->info, p1, hw->samples);
-
- fail:
- status = FSOUND_Sample_Unlock (fmd->fmod_sample, p1, p2, len1, len2);
- if (!status) {
- fmod_logerr ("Failed to unlock sample\n");
- }
-}
-
-static void fmod_write_sample (HWVoiceOut *hw, uint8_t *dst, int dst_len)
-{
- int src_len1 = dst_len;
- int src_len2 = 0;
- int pos = hw->rpos + dst_len;
- struct st_sample *src1 = hw->mix_buf + hw->rpos;
- struct st_sample *src2 = NULL;
-
- if (pos > hw->samples) {
- src_len1 = hw->samples - hw->rpos;
- src2 = hw->mix_buf;
- src_len2 = dst_len - src_len1;
- pos = src_len2;
- }
-
- if (src_len1) {
- hw->clip (dst, src1, src_len1);
- }
-
- if (src_len2) {
- dst = advance (dst, src_len1 << hw->info.shift);
- hw->clip (dst, src2, src_len2);
- }
-
- hw->rpos = pos % hw->samples;
-}
-
-static int fmod_unlock_sample (FSOUND_SAMPLE *sample, void *p1, void *p2,
- unsigned int blen1, unsigned int blen2)
-{
- int status = FSOUND_Sample_Unlock (sample, p1, p2, blen1, blen2);
- if (!status) {
- fmod_logerr ("Failed to unlock sample\n");
- return -1;
- }
- return 0;
-}
-
-static int fmod_lock_sample (
- FSOUND_SAMPLE *sample,
- struct audio_pcm_info *info,
- int pos,
- int len,
- void **p1,
- void **p2,
- unsigned int *blen1,
- unsigned int *blen2
- )
-{
- int status;
-
- status = FSOUND_Sample_Lock (
- sample,
- pos << info->shift,
- len << info->shift,
- p1,
- p2,
- blen1,
- blen2
- );
-
- if (!status) {
- fmod_logerr ("Failed to lock sample\n");
- return -1;
- }
-
- if ((*blen1 & info->align) || (*blen2 & info->align)) {
- dolog ("Lock returned misaligned length %d, %d, alignment %d\n",
- *blen1, *blen2, info->align + 1);
-
- fmod_unlock_sample (sample, *p1, *p2, *blen1, *blen2);
-
- *p1 = NULL - 1;
- *p2 = NULL - 1;
- *blen1 = ~0U;
- *blen2 = ~0U;
- return -1;
- }
-
- if (!*p1 && *blen1) {
- dolog ("warning: !p1 && blen1=%d\n", *blen1);
- *blen1 = 0;
- }
-
- if (!p2 && *blen2) {
- dolog ("warning: !p2 && blen2=%d\n", *blen2);
- *blen2 = 0;
- }
-
- return 0;
-}
-
-static int fmod_run_out (HWVoiceOut *hw, int live)
-{
- FMODVoiceOut *fmd = (FMODVoiceOut *) hw;
- int decr;
- void *p1 = 0, *p2 = 0;
- unsigned int blen1 = 0, blen2 = 0;
- unsigned int len1 = 0, len2 = 0;
-
- if (!hw->pending_disable) {
- return 0;
- }
-
- decr = live;
-
- if (fmd->channel >= 0) {
- int len = decr;
- int old_pos = fmd->old_pos;
- int ppos = FSOUND_GetCurrentPosition (fmd->channel);
-
- if (ppos == old_pos || !ppos) {
- return 0;
- }
-
- if ((old_pos < ppos) && ((old_pos + len) > ppos)) {
- len = ppos - old_pos;
- }
- else {
- if ((old_pos > ppos) && ((old_pos + len) > (ppos + hw->samples))) {
- len = hw->samples - old_pos + ppos;
- }
- }
- decr = len;
-
- if (audio_bug (AUDIO_FUNC, decr < 0)) {
- dolog ("decr=%d live=%d ppos=%d old_pos=%d len=%d\n",
- decr, live, ppos, old_pos, len);
- return 0;
- }
- }
-
-
- if (!decr) {
- return 0;
- }
-
- if (fmod_lock_sample (fmd->fmod_sample, &fmd->hw.info,
- fmd->old_pos, decr,
- &p1, &p2,
- &blen1, &blen2)) {
- return 0;
- }
-
- len1 = blen1 >> hw->info.shift;
- len2 = blen2 >> hw->info.shift;
- ldebug ("%p %p %d %d %d %d\n", p1, p2, len1, len2, blen1, blen2);
- decr = len1 + len2;
-
- if (p1 && len1) {
- fmod_write_sample (hw, p1, len1);
- }
-
- if (p2 && len2) {
- fmod_write_sample (hw, p2, len2);
- }
-
- fmod_unlock_sample (fmd->fmod_sample, p1, p2, blen1, blen2);
-
- fmd->old_pos = (fmd->old_pos + decr) % hw->samples;
- return decr;
-}
-
-static int aud_to_fmodfmt (audfmt_e fmt, int stereo)
-{
- int mode = FSOUND_LOOP_NORMAL;
-
- switch (fmt) {
- case AUD_FMT_S8:
- mode |= FSOUND_SIGNED | FSOUND_8BITS;
- break;
-
- case AUD_FMT_U8:
- mode |= FSOUND_UNSIGNED | FSOUND_8BITS;
- break;
-
- case AUD_FMT_S16:
- mode |= FSOUND_SIGNED | FSOUND_16BITS;
- break;
-
- case AUD_FMT_U16:
- mode |= FSOUND_UNSIGNED | FSOUND_16BITS;
- break;
-
- default:
- dolog ("Internal logic error: Bad audio format %d\n", fmt);
-#ifdef DEBUG_FMOD
- abort ();
-#endif
- mode |= FSOUND_8BITS;
- }
- mode |= stereo ? FSOUND_STEREO : FSOUND_MONO;
- return mode;
-}
-
-static void fmod_fini_out (HWVoiceOut *hw)
-{
- FMODVoiceOut *fmd = (FMODVoiceOut *) hw;
-
- if (fmd->fmod_sample) {
- FSOUND_Sample_Free (fmd->fmod_sample);
- fmd->fmod_sample = 0;
-
- if (fmd->channel >= 0) {
- FSOUND_StopSound (fmd->channel);
- }
- }
-}
-
-static int fmod_init_out (HWVoiceOut *hw, struct audsettings *as)
-{
- int mode, channel;
- FMODVoiceOut *fmd = (FMODVoiceOut *) hw;
- struct audsettings obt_as = *as;
-
- mode = aud_to_fmodfmt (as->fmt, as->nchannels == 2 ? 1 : 0);
- fmd->fmod_sample = FSOUND_Sample_Alloc (
- FSOUND_FREE, /* index */
- conf.nb_samples, /* length */
- mode, /* mode */
- as->freq, /* freq */
- 255, /* volume */
- 128, /* pan */
- 255 /* priority */
- );
-
- if (!fmd->fmod_sample) {
- fmod_logerr2 ("DAC", "Failed to allocate FMOD sample\n");
- return -1;
- }
-
- channel = FSOUND_PlaySoundEx (FSOUND_FREE, fmd->fmod_sample, 0, 1);
- if (channel < 0) {
- fmod_logerr2 ("DAC", "Failed to start playing sound\n");
- FSOUND_Sample_Free (fmd->fmod_sample);
- return -1;
- }
- fmd->channel = channel;
-
- /* FMOD always operates on little endian frames? */
- obt_as.endianness = 0;
- audio_pcm_init_info (&hw->info, &obt_as);
- hw->samples = conf.nb_samples;
- return 0;
-}
-
-static int fmod_ctl_out (HWVoiceOut *hw, int cmd, ...)
-{
- int status;
- FMODVoiceOut *fmd = (FMODVoiceOut *) hw;
-
- switch (cmd) {
- case VOICE_ENABLE:
- fmod_clear_sample (fmd);
- status = FSOUND_SetPaused (fmd->channel, 0);
- if (!status) {
- fmod_logerr ("Failed to resume channel %d\n", fmd->channel);
- }
- break;
-
- case VOICE_DISABLE:
- status = FSOUND_SetPaused (fmd->channel, 1);
- if (!status) {
- fmod_logerr ("Failed to pause channel %d\n", fmd->channel);
- }
- break;
- }
- return 0;
-}
-
-static int fmod_init_in (HWVoiceIn *hw, struct audsettings *as)
-{
- int mode;
- FMODVoiceIn *fmd = (FMODVoiceIn *) hw;
- struct audsettings obt_as = *as;
-
- if (conf.broken_adc) {
- return -1;
- }
-
- mode = aud_to_fmodfmt (as->fmt, as->nchannels == 2 ? 1 : 0);
- fmd->fmod_sample = FSOUND_Sample_Alloc (
- FSOUND_FREE, /* index */
- conf.nb_samples, /* length */
- mode, /* mode */
- as->freq, /* freq */
- 255, /* volume */
- 128, /* pan */
- 255 /* priority */
- );
-
- if (!fmd->fmod_sample) {
- fmod_logerr2 ("ADC", "Failed to allocate FMOD sample\n");
- return -1;
- }
-
- /* FMOD always operates on little endian frames? */
- obt_as.endianness = 0;
- audio_pcm_init_info (&hw->info, &obt_as);
- hw->samples = conf.nb_samples;
- return 0;
-}
-
-static void fmod_fini_in (HWVoiceIn *hw)
-{
- FMODVoiceIn *fmd = (FMODVoiceIn *) hw;
-
- if (fmd->fmod_sample) {
- FSOUND_Record_Stop ();
- FSOUND_Sample_Free (fmd->fmod_sample);
- fmd->fmod_sample = 0;
- }
-}
-
-static int fmod_run_in (HWVoiceIn *hw)
-{
- FMODVoiceIn *fmd = (FMODVoiceIn *) hw;
- int hwshift = hw->info.shift;
- int live, dead, new_pos, len;
- unsigned int blen1 = 0, blen2 = 0;
- unsigned int len1, len2;
- unsigned int decr;
- void *p1, *p2;
-
- live = audio_pcm_hw_get_live_in (hw);
- dead = hw->samples - live;
- if (!dead) {
- return 0;
- }
-
- new_pos = FSOUND_Record_GetPosition ();
- if (new_pos < 0) {
- fmod_logerr ("Could not get recording position\n");
- return 0;
- }
-
- len = audio_ring_dist (new_pos, hw->wpos, hw->samples);
- if (!len) {
- return 0;
- }
- len = audio_MIN (len, dead);
-
- if (fmod_lock_sample (fmd->fmod_sample, &fmd->hw.info,
- hw->wpos, len,
- &p1, &p2,
- &blen1, &blen2)) {
- return 0;
- }
-
- len1 = blen1 >> hwshift;
- len2 = blen2 >> hwshift;
- decr = len1 + len2;
-
- if (p1 && blen1) {
- hw->conv (hw->conv_buf + hw->wpos, p1, len1);
- }
- if (p2 && len2) {
- hw->conv (hw->conv_buf, p2, len2);
- }
-
- fmod_unlock_sample (fmd->fmod_sample, p1, p2, blen1, blen2);
- hw->wpos = (hw->wpos + decr) % hw->samples;
- return decr;
-}
-
-static struct {
- const char *name;
- int type;
-} drvtab[] = {
- { .name = "none", .type = FSOUND_OUTPUT_NOSOUND },
-#ifdef _WIN32
- { .name = "winmm", .type = FSOUND_OUTPUT_WINMM },
- { .name = "dsound", .type = FSOUND_OUTPUT_DSOUND },
- { .name = "a3d", .type = FSOUND_OUTPUT_A3D },
- { .name = "asio", .type = FSOUND_OUTPUT_ASIO },
-#endif
-#ifdef __linux__
- { .name = "oss", .type = FSOUND_OUTPUT_OSS },
- { .name = "alsa", .type = FSOUND_OUTPUT_ALSA },
- { .name = "esd", .type = FSOUND_OUTPUT_ESD },
-#endif
-#ifdef __APPLE__
- { .name = "mac", .type = FSOUND_OUTPUT_MAC },
-#endif
-#if 0
- { .name = "xbox", .type = FSOUND_OUTPUT_XBOX },
- { .name = "ps2", .type = FSOUND_OUTPUT_PS2 },
- { .name = "gcube", .type = FSOUND_OUTPUT_GC },
-#endif
- { .name = "none-realtime", .type = FSOUND_OUTPUT_NOSOUND_NONREALTIME }
-};
-
-static void *fmod_audio_init (void)
-{
- size_t i;
- double ver;
- int status;
- int output_type = -1;
- const char *drv = conf.drvname;
-
- ver = FSOUND_GetVersion ();
- if (ver < FMOD_VERSION) {
- dolog ("Wrong FMOD version %f, need at least %f\n", ver, FMOD_VERSION);
- return NULL;
- }
-
-#ifdef __linux__
- if (ver < 3.75) {
- dolog ("FMOD before 3.75 has bug preventing ADC from working\n"
- "ADC will be disabled.\n");
- conf.broken_adc = 1;
- }
-#endif
-
- if (drv) {
- int found = 0;
- for (i = 0; i < ARRAY_SIZE (drvtab); i++) {
- if (!strcmp (drv, drvtab[i].name)) {
- output_type = drvtab[i].type;
- found = 1;
- break;
- }
- }
- if (!found) {
- dolog ("Unknown FMOD driver `%s'\n", drv);
- dolog ("Valid drivers:\n");
- for (i = 0; i < ARRAY_SIZE (drvtab); i++) {
- dolog (" %s\n", drvtab[i].name);
- }
- }
- }
-
- if (output_type != -1) {
- status = FSOUND_SetOutput (output_type);
- if (!status) {
- fmod_logerr ("FSOUND_SetOutput(%d) failed\n", output_type);
- return NULL;
- }
- }
-
- if (conf.bufsize) {
- status = FSOUND_SetBufferSize (conf.bufsize);
- if (!status) {
- fmod_logerr ("FSOUND_SetBufferSize (%d) failed\n", conf.bufsize);
- }
- }
-
- status = FSOUND_Init (conf.freq, conf.nb_channels, 0);
- if (!status) {
- fmod_logerr ("FSOUND_Init failed\n");
- return NULL;
- }
-
- return &conf;
-}
-
-static int fmod_read (SWVoiceIn *sw, void *buf, int size)
-{
- return audio_pcm_sw_read (sw, buf, size);
-}
-
-static int fmod_ctl_in (HWVoiceIn *hw, int cmd, ...)
-{
- int status;
- FMODVoiceIn *fmd = (FMODVoiceIn *) hw;
-
- switch (cmd) {
- case VOICE_ENABLE:
- status = FSOUND_Record_StartSample (fmd->fmod_sample, 1);
- if (!status) {
- fmod_logerr ("Failed to start recording\n");
- }
- break;
-
- case VOICE_DISABLE:
- status = FSOUND_Record_Stop ();
- if (!status) {
- fmod_logerr ("Failed to stop recording\n");
- }
- break;
- }
- return 0;
-}
-
-static void fmod_audio_fini (void *opaque)
-{
- (void) opaque;
- FSOUND_Close ();
-}
-
-static struct audio_option fmod_options[] = {
- {
- .name = "DRV",
- .tag = AUD_OPT_STR,
- .valp = &conf.drvname,
- .descr = "FMOD driver"
- },
- {
- .name = "FREQ",
- .tag = AUD_OPT_INT,
- .valp = &conf.freq,
- .descr = "Default frequency"
- },
- {
- .name = "SAMPLES",
- .tag = AUD_OPT_INT,
- .valp = &conf.nb_samples,
- .descr = "Buffer size in samples"
- },
- {
- .name = "CHANNELS",
- .tag = AUD_OPT_INT,
- .valp = &conf.nb_channels,
- .descr = "Number of default channels (1 - mono, 2 - stereo)"
- },
- {
- .name = "BUFSIZE",
- .tag = AUD_OPT_INT,
- .valp = &conf.bufsize,
- .descr = "(undocumented)"
- },
- { /* End of list */ }
-};
-
-static struct audio_pcm_ops fmod_pcm_ops = {
- .init_out = fmod_init_out,
- .fini_out = fmod_fini_out,
- .run_out = fmod_run_out,
- .write = fmod_write,
- .ctl_out = fmod_ctl_out,
-
- .init_in = fmod_init_in,
- .fini_in = fmod_fini_in,
- .run_in = fmod_run_in,
- .read = fmod_read,
- .ctl_in = fmod_ctl_in
-};
-
-struct audio_driver fmod_audio_driver = {
- .name = "fmod",
- .descr = "FMOD 3.xx http://www.fmod.org",
- .options = fmod_options,
- .init = fmod_audio_init,
- .fini = fmod_audio_fini,
- .pcm_ops = &fmod_pcm_ops,
- .can_be_default = 1,
- .max_voices_out = INT_MAX,
- .max_voices_in = INT_MAX,
- .voice_size_out = sizeof (FMODVoiceOut),
- .voice_size_in = sizeof (FMODVoiceIn)
-};
diff --git a/configure b/configure
index acd471d..cdece43 100755
--- a/configure
+++ b/configure
@@ -285,8 +285,6 @@ sysconfdir="\${prefix}/etc"
local_statedir="\${prefix}/var"
confsuffix="/qemu"
slirp="yes"
-fmod_lib=""
-fmod_inc=""
oss_lib=""
bsd="no"
linux="no"
@@ -569,7 +567,7 @@ CYGWIN*)
;;
MINGW32*)
mingw32="yes"
- audio_possible_drivers="winwave dsound sdl fmod"
+ audio_possible_drivers="winwave dsound sdl"
audio_drv_list="winwave"
;;
GNU/kFreeBSD)
@@ -619,7 +617,7 @@ Darwin)
fi
cocoa="yes"
audio_drv_list="coreaudio"
- audio_possible_drivers="coreaudio sdl fmod"
+ audio_possible_drivers="coreaudio sdl"
LDFLAGS="-framework CoreFoundation -framework IOKit $LDFLAGS"
libs_softmmu="-F/System/Library/Frameworks -framework Cocoa -framework IOKit $libs_softmmu"
# Disable attempts to use ObjectiveC features in os/object.h since they
@@ -680,9 +678,6 @@ Haiku)
kvm="yes"
vhost_net="yes"
vhost_scsi="yes"
- if [ "$cpu" = "i386" -o "$cpu" = "x86_64" -o "$cpu" = "x32" ] ; then
- audio_possible_drivers="$audio_possible_drivers fmod"
- fi
QEMU_INCLUDES="-I\$(SRC_PATH)/linux-headers -I$(pwd)/linux-headers $QEMU_INCLUDES"
;;
esac
@@ -847,10 +842,6 @@ for opt do
;;
--enable-vnc) vnc="yes"
;;
- --fmod-lib=*) fmod_lib="$optarg"
- ;;
- --fmod-inc=*) fmod_inc="$optarg"
- ;;
--oss-lib=*) oss_lib="$optarg"
;;
--audio-drv-list=*) audio_drv_list="$optarg"
@@ -1349,8 +1340,6 @@ Advanced options (experts only):
--disable-guest-base disable GUEST_BASE support
--enable-pie build Position Independent Executables
--disable-pie do not build Position Independent Executables
- --fmod-lib path to FMOD library
- --fmod-inc path to FMOD includes
--oss-lib path to OSS library
--cpu=CPU Build for host CPU [$cpu]
--disable-uuid disable uuid support
@@ -2621,15 +2610,6 @@ for drv in $audio_drv_list; do
libs_softmmu="-lasound $libs_softmmu"
;;
- fmod)
- if test -z $fmod_lib || test -z $fmod_inc; then
- error_exit "You must specify path to FMOD library and headers" \
- "Example: --fmod-inc=/path/include/fmod --fmod-lib=/path/lib/libfmod-3.74.so"
- fi
- audio_drv_probe $drv fmod.h $fmod_lib "return FSOUND_GetVersion();" "-I $fmod_inc"
- libs_softmmu="$fmod_lib $libs_softmmu"
- ;;
-
pa)
audio_drv_probe $drv pulse/mainloop.h "-lpulse" \
"pa_mainloop *m = 0; pa_mainloop_free (m); return 0;"
@@ -4621,9 +4601,6 @@ echo "CONFIG_AUDIO_DRIVERS=$audio_drv_list" >> $config_host_mak
for drv in $audio_drv_list; do
def=CONFIG_`echo $drv | LC_ALL=C tr '[a-z]' '[A-Z]'`
echo "$def=y" >> $config_host_mak
- if test "$drv" = "fmod"; then
- echo "FMOD_CFLAGS=-I$fmod_inc" >> $config_host_mak
- fi
done
if test "$audio_pt_int" = "yes" ; then
echo "CONFIG_AUDIO_PT_INT=y" >> $config_host_mak
--
2.4.2
^ permalink raw reply related [flat|nested] 26+ messages in thread* [Qemu-devel] [PATCH 03/12] audio: remove sdl backend
2015-06-01 13:23 [Qemu-devel] [PATCH 00/12] Audio backend cleanup Kővágó, Zoltán
2015-06-01 13:23 ` [Qemu-devel] [PATCH 01/12] audio: remove esd backend Kővágó, Zoltán
2015-06-01 13:23 ` [Qemu-devel] [PATCH 02/12] audio: remove fmod backend Kővágó, Zoltán
@ 2015-06-01 13:23 ` Kővágó, Zoltán
2015-06-01 13:23 ` [Qemu-devel] [PATCH 04/12] audio: remove winwave audio driver Kővágó, Zoltán
` (9 subsequent siblings)
12 siblings, 0 replies; 26+ messages in thread
From: Kővágó, Zoltán @ 2015-06-01 13:23 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
It's broken and we have native drivers for almost all platforms.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/Makefile.objs | 3 -
audio/audio_int.h | 1 -
audio/sdlaudio.c | 458 ----------------------------------------------------
configure | 24 ++-
4 files changed, 11 insertions(+), 475 deletions(-)
delete mode 100644 audio/sdlaudio.c
diff --git a/audio/Makefile.objs b/audio/Makefile.objs
index b4c0608..3c84972 100644
--- a/audio/Makefile.objs
+++ b/audio/Makefile.objs
@@ -1,5 +1,4 @@
common-obj-y = audio.o noaudio.o wavaudio.o mixeng.o
-common-obj-$(CONFIG_SDL) += sdlaudio.o
common-obj-$(CONFIG_OSS) += ossaudio.o
common-obj-$(CONFIG_SPICE) += spiceaudio.o
common-obj-$(CONFIG_COREAUDIO) += coreaudio.o
@@ -10,5 +9,3 @@ common-obj-$(CONFIG_WINWAVE) += winwaveaudio.o
common-obj-$(CONFIG_AUDIO_PT_INT) += audio_pt_int.o
common-obj-$(CONFIG_AUDIO_WIN_INT) += audio_win_int.o
common-obj-y += wavcapture.o
-
-sdlaudio.o-cflags := $(SDL_CFLAGS)
diff --git a/audio/audio_int.h b/audio/audio_int.h
index 7445602..d593879 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -204,7 +204,6 @@ struct AudioState {
extern struct audio_driver no_audio_driver;
extern struct audio_driver oss_audio_driver;
-extern struct audio_driver sdl_audio_driver;
extern struct audio_driver wav_audio_driver;
extern struct audio_driver alsa_audio_driver;
extern struct audio_driver coreaudio_audio_driver;
diff --git a/audio/sdlaudio.c b/audio/sdlaudio.c
deleted file mode 100644
index d24daa5..0000000
--- a/audio/sdlaudio.c
+++ /dev/null
@@ -1,458 +0,0 @@
-/*
- * QEMU SDL audio driver
- *
- * Copyright (c) 2004-2005 Vassili Karpov (malc)
- *
- * Permission is hereby granted, free of charge, to any person obtaining a copy
- * of this software and associated documentation files (the "Software"), to deal
- * in the Software without restriction, including without limitation the rights
- * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
- * copies of the Software, and to permit persons to whom the Software is
- * furnished to do so, subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be included in
- * all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
- * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
- * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
- * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
- * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
- * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
- * THE SOFTWARE.
- */
-#include <SDL.h>
-#include <SDL_thread.h>
-#include "qemu-common.h"
-#include "audio.h"
-
-#ifndef _WIN32
-#ifdef __sun__
-#define _POSIX_PTHREAD_SEMANTICS 1
-#elif defined(__OpenBSD__) || defined(__FreeBSD__) || defined(__DragonFly__)
-#include <pthread.h>
-#endif
-#endif
-
-#define AUDIO_CAP "sdl"
-#include "audio_int.h"
-
-typedef struct SDLVoiceOut {
- HWVoiceOut hw;
- int live;
- int rpos;
- int decr;
-} SDLVoiceOut;
-
-static struct {
- int nb_samples;
-} conf = {
- .nb_samples = 1024
-};
-
-static struct SDLAudioState {
- int exit;
- SDL_mutex *mutex;
- SDL_sem *sem;
- int initialized;
-} glob_sdl;
-typedef struct SDLAudioState SDLAudioState;
-
-static void GCC_FMT_ATTR (1, 2) sdl_logerr (const char *fmt, ...)
-{
- va_list ap;
-
- va_start (ap, fmt);
- AUD_vlog (AUDIO_CAP, fmt, ap);
- va_end (ap);
-
- AUD_log (AUDIO_CAP, "Reason: %s\n", SDL_GetError ());
-}
-
-static int sdl_lock (SDLAudioState *s, const char *forfn)
-{
- if (SDL_LockMutex (s->mutex)) {
- sdl_logerr ("SDL_LockMutex for %s failed\n", forfn);
- return -1;
- }
- return 0;
-}
-
-static int sdl_unlock (SDLAudioState *s, const char *forfn)
-{
- if (SDL_UnlockMutex (s->mutex)) {
- sdl_logerr ("SDL_UnlockMutex for %s failed\n", forfn);
- return -1;
- }
- return 0;
-}
-
-static int sdl_post (SDLAudioState *s, const char *forfn)
-{
- if (SDL_SemPost (s->sem)) {
- sdl_logerr ("SDL_SemPost for %s failed\n", forfn);
- return -1;
- }
- return 0;
-}
-
-static int sdl_wait (SDLAudioState *s, const char *forfn)
-{
- if (SDL_SemWait (s->sem)) {
- sdl_logerr ("SDL_SemWait for %s failed\n", forfn);
- return -1;
- }
- return 0;
-}
-
-static int sdl_unlock_and_post (SDLAudioState *s, const char *forfn)
-{
- if (sdl_unlock (s, forfn)) {
- return -1;
- }
-
- return sdl_post (s, forfn);
-}
-
-static int aud_to_sdlfmt (audfmt_e fmt)
-{
- switch (fmt) {
- case AUD_FMT_S8:
- return AUDIO_S8;
-
- case AUD_FMT_U8:
- return AUDIO_U8;
-
- case AUD_FMT_S16:
- return AUDIO_S16LSB;
-
- case AUD_FMT_U16:
- return AUDIO_U16LSB;
-
- default:
- dolog ("Internal logic error: Bad audio format %d\n", fmt);
-#ifdef DEBUG_AUDIO
- abort ();
-#endif
- return AUDIO_U8;
- }
-}
-
-static int sdl_to_audfmt(int sdlfmt, audfmt_e *fmt, int *endianness)
-{
- switch (sdlfmt) {
- case AUDIO_S8:
- *endianness = 0;
- *fmt = AUD_FMT_S8;
- break;
-
- case AUDIO_U8:
- *endianness = 0;
- *fmt = AUD_FMT_U8;
- break;
-
- case AUDIO_S16LSB:
- *endianness = 0;
- *fmt = AUD_FMT_S16;
- break;
-
- case AUDIO_U16LSB:
- *endianness = 0;
- *fmt = AUD_FMT_U16;
- break;
-
- case AUDIO_S16MSB:
- *endianness = 1;
- *fmt = AUD_FMT_S16;
- break;
-
- case AUDIO_U16MSB:
- *endianness = 1;
- *fmt = AUD_FMT_U16;
- break;
-
- default:
- dolog ("Unrecognized SDL audio format %d\n", sdlfmt);
- return -1;
- }
-
- return 0;
-}
-
-static int sdl_open (SDL_AudioSpec *req, SDL_AudioSpec *obt)
-{
- int status;
-#ifndef _WIN32
- int err;
- sigset_t new, old;
-
- /* Make sure potential threads created by SDL don't hog signals. */
- err = sigfillset (&new);
- if (err) {
- dolog ("sdl_open: sigfillset failed: %s\n", strerror (errno));
- return -1;
- }
- err = pthread_sigmask (SIG_BLOCK, &new, &old);
- if (err) {
- dolog ("sdl_open: pthread_sigmask failed: %s\n", strerror (err));
- return -1;
- }
-#endif
-
- status = SDL_OpenAudio (req, obt);
- if (status) {
- sdl_logerr ("SDL_OpenAudio failed\n");
- }
-
-#ifndef _WIN32
- err = pthread_sigmask (SIG_SETMASK, &old, NULL);
- if (err) {
- dolog ("sdl_open: pthread_sigmask (restore) failed: %s\n",
- strerror (errno));
- /* We have failed to restore original signal mask, all bets are off,
- so exit the process */
- exit (EXIT_FAILURE);
- }
-#endif
- return status;
-}
-
-static void sdl_close (SDLAudioState *s)
-{
- if (s->initialized) {
- sdl_lock (s, "sdl_close");
- s->exit = 1;
- sdl_unlock_and_post (s, "sdl_close");
- SDL_PauseAudio (1);
- SDL_CloseAudio ();
- s->initialized = 0;
- }
-}
-
-static void sdl_callback (void *opaque, Uint8 *buf, int len)
-{
- SDLVoiceOut *sdl = opaque;
- SDLAudioState *s = &glob_sdl;
- HWVoiceOut *hw = &sdl->hw;
- int samples = len >> hw->info.shift;
-
- if (s->exit) {
- return;
- }
-
- while (samples) {
- int to_mix, decr;
-
- /* dolog ("in callback samples=%d\n", samples); */
- sdl_wait (s, "sdl_callback");
- if (s->exit) {
- return;
- }
-
- if (sdl_lock (s, "sdl_callback")) {
- return;
- }
-
- if (audio_bug (AUDIO_FUNC, sdl->live < 0 || sdl->live > hw->samples)) {
- dolog ("sdl->live=%d hw->samples=%d\n",
- sdl->live, hw->samples);
- return;
- }
-
- if (!sdl->live) {
- goto again;
- }
-
- /* dolog ("in callback live=%d\n", live); */
- to_mix = audio_MIN (samples, sdl->live);
- decr = to_mix;
- while (to_mix) {
- int chunk = audio_MIN (to_mix, hw->samples - hw->rpos);
- struct st_sample *src = hw->mix_buf + hw->rpos;
-
- /* dolog ("in callback to_mix %d, chunk %d\n", to_mix, chunk); */
- hw->clip (buf, src, chunk);
- sdl->rpos = (sdl->rpos + chunk) % hw->samples;
- to_mix -= chunk;
- buf += chunk << hw->info.shift;
- }
- samples -= decr;
- sdl->live -= decr;
- sdl->decr += decr;
-
- again:
- if (sdl_unlock (s, "sdl_callback")) {
- return;
- }
- }
- /* dolog ("done len=%d\n", len); */
-}
-
-static int sdl_write_out (SWVoiceOut *sw, void *buf, int len)
-{
- return audio_pcm_sw_write (sw, buf, len);
-}
-
-static int sdl_run_out (HWVoiceOut *hw, int live)
-{
- int decr;
- SDLVoiceOut *sdl = (SDLVoiceOut *) hw;
- SDLAudioState *s = &glob_sdl;
-
- if (sdl_lock (s, "sdl_run_out")) {
- return 0;
- }
-
- if (sdl->decr > live) {
- ldebug ("sdl->decr %d live %d sdl->live %d\n",
- sdl->decr,
- live,
- sdl->live);
- }
-
- decr = audio_MIN (sdl->decr, live);
- sdl->decr -= decr;
-
- sdl->live = live - decr;
- hw->rpos = sdl->rpos;
-
- if (sdl->live > 0) {
- sdl_unlock_and_post (s, "sdl_run_out");
- }
- else {
- sdl_unlock (s, "sdl_run_out");
- }
- return decr;
-}
-
-static void sdl_fini_out (HWVoiceOut *hw)
-{
- (void) hw;
-
- sdl_close (&glob_sdl);
-}
-
-static int sdl_init_out (HWVoiceOut *hw, struct audsettings *as)
-{
- SDLVoiceOut *sdl = (SDLVoiceOut *) hw;
- SDLAudioState *s = &glob_sdl;
- SDL_AudioSpec req, obt;
- int endianness;
- int err;
- audfmt_e effective_fmt;
- struct audsettings obt_as;
-
- req.freq = as->freq;
- req.format = aud_to_sdlfmt (as->fmt);
- req.channels = as->nchannels;
- req.samples = conf.nb_samples;
- req.callback = sdl_callback;
- req.userdata = sdl;
-
- if (sdl_open (&req, &obt)) {
- return -1;
- }
-
- err = sdl_to_audfmt(obt.format, &effective_fmt, &endianness);
- if (err) {
- sdl_close (s);
- return -1;
- }
-
- obt_as.freq = obt.freq;
- obt_as.nchannels = obt.channels;
- obt_as.fmt = effective_fmt;
- obt_as.endianness = endianness;
-
- audio_pcm_init_info (&hw->info, &obt_as);
- hw->samples = obt.samples;
-
- s->initialized = 1;
- s->exit = 0;
- SDL_PauseAudio (0);
- return 0;
-}
-
-static int sdl_ctl_out (HWVoiceOut *hw, int cmd, ...)
-{
- (void) hw;
-
- switch (cmd) {
- case VOICE_ENABLE:
- SDL_PauseAudio (0);
- break;
-
- case VOICE_DISABLE:
- SDL_PauseAudio (1);
- break;
- }
- return 0;
-}
-
-static void *sdl_audio_init (void)
-{
- SDLAudioState *s = &glob_sdl;
-
- if (SDL_InitSubSystem (SDL_INIT_AUDIO)) {
- sdl_logerr ("SDL failed to initialize audio subsystem\n");
- return NULL;
- }
-
- s->mutex = SDL_CreateMutex ();
- if (!s->mutex) {
- sdl_logerr ("Failed to create SDL mutex\n");
- SDL_QuitSubSystem (SDL_INIT_AUDIO);
- return NULL;
- }
-
- s->sem = SDL_CreateSemaphore (0);
- if (!s->sem) {
- sdl_logerr ("Failed to create SDL semaphore\n");
- SDL_DestroyMutex (s->mutex);
- SDL_QuitSubSystem (SDL_INIT_AUDIO);
- return NULL;
- }
-
- return s;
-}
-
-static void sdl_audio_fini (void *opaque)
-{
- SDLAudioState *s = opaque;
- sdl_close (s);
- SDL_DestroySemaphore (s->sem);
- SDL_DestroyMutex (s->mutex);
- SDL_QuitSubSystem (SDL_INIT_AUDIO);
-}
-
-static struct audio_option sdl_options[] = {
- {
- .name = "SAMPLES",
- .tag = AUD_OPT_INT,
- .valp = &conf.nb_samples,
- .descr = "Size of SDL buffer in samples"
- },
- { /* End of list */ }
-};
-
-static struct audio_pcm_ops sdl_pcm_ops = {
- .init_out = sdl_init_out,
- .fini_out = sdl_fini_out,
- .run_out = sdl_run_out,
- .write = sdl_write_out,
- .ctl_out = sdl_ctl_out,
-};
-
-struct audio_driver sdl_audio_driver = {
- .name = "sdl",
- .descr = "SDL http://www.libsdl.org",
- .options = sdl_options,
- .init = sdl_audio_init,
- .fini = sdl_audio_fini,
- .pcm_ops = &sdl_pcm_ops,
- .can_be_default = 1,
- .max_voices_out = 1,
- .max_voices_in = 0,
- .voice_size_out = sizeof (SDLVoiceOut),
- .voice_size_in = 0
-};
diff --git a/configure b/configure
index cdece43..154ecbe 100755
--- a/configure
+++ b/configure
@@ -562,24 +562,24 @@ case $targetos in
CYGWIN*)
mingw32="yes"
QEMU_CFLAGS="-mno-cygwin $QEMU_CFLAGS"
- audio_possible_drivers="winwave sdl"
+ audio_possible_drivers="winwave"
audio_drv_list="winwave"
;;
MINGW32*)
mingw32="yes"
- audio_possible_drivers="winwave dsound sdl"
+ audio_possible_drivers="winwave dsound"
audio_drv_list="winwave"
;;
GNU/kFreeBSD)
bsd="yes"
audio_drv_list="oss"
- audio_possible_drivers="oss sdl pa"
+ audio_possible_drivers="oss pa"
;;
FreeBSD)
bsd="yes"
make="${MAKE-gmake}"
audio_drv_list="oss"
- audio_possible_drivers="oss sdl pa"
+ audio_possible_drivers="oss pa"
# needed for kinfo_getvmmap(3) in libutil.h
LIBS="-lutil $LIBS"
netmap="" # enable netmap autodetect
@@ -589,22 +589,20 @@ DragonFly)
bsd="yes"
make="${MAKE-gmake}"
audio_drv_list="oss"
- audio_possible_drivers="oss sdl pa"
+ audio_possible_drivers="oss pa"
HOST_VARIANT_DIR="dragonfly"
;;
NetBSD)
bsd="yes"
make="${MAKE-gmake}"
audio_drv_list="oss"
- audio_possible_drivers="oss sdl"
+ audio_possible_drivers="oss"
oss_lib="-lossaudio"
HOST_VARIANT_DIR="netbsd"
;;
OpenBSD)
bsd="yes"
make="${MAKE-gmake}"
- audio_drv_list="sdl"
- audio_possible_drivers="sdl"
HOST_VARIANT_DIR="openbsd"
;;
Darwin)
@@ -617,7 +615,7 @@ Darwin)
fi
cocoa="yes"
audio_drv_list="coreaudio"
- audio_possible_drivers="coreaudio sdl"
+ audio_possible_drivers="coreaudio"
LDFLAGS="-framework CoreFoundation -framework IOKit $LDFLAGS"
libs_softmmu="-F/System/Library/Frameworks -framework Cocoa -framework IOKit $libs_softmmu"
# Disable attempts to use ObjectiveC features in os/object.h since they
@@ -651,7 +649,7 @@ SunOS)
if test -f /usr/include/sys/soundcard.h ; then
audio_drv_list="oss"
fi
- audio_possible_drivers="oss sdl"
+ audio_possible_drivers="oss"
# needed for CMSG_ macros in sys/socket.h
QEMU_CFLAGS="-D_XOPEN_SOURCE=600 $QEMU_CFLAGS"
# needed for TIOCWIN* defines in termios.h
@@ -672,7 +670,7 @@ Haiku)
;;
*)
audio_drv_list="oss"
- audio_possible_drivers="oss alsa sdl pa"
+ audio_possible_drivers="oss alsa pa"
linux="yes"
linux_user="yes"
kvm="yes"
@@ -2630,8 +2628,8 @@ for drv in $audio_drv_list; do
libs_softmmu="$oss_lib $libs_softmmu"
;;
- sdl|wav)
- # XXX: Probes for CoreAudio, DirectSound, SDL(?)
+ wav)
+ # XXX: Probes for CoreAudio, DirectSound
;;
winwave)
--
2.4.2
^ permalink raw reply related [flat|nested] 26+ messages in thread* [Qemu-devel] [PATCH 04/12] audio: remove winwave audio driver
2015-06-01 13:23 [Qemu-devel] [PATCH 00/12] Audio backend cleanup Kővágó, Zoltán
` (2 preceding siblings ...)
2015-06-01 13:23 ` [Qemu-devel] [PATCH 03/12] audio: remove sdl backend Kővágó, Zoltán
@ 2015-06-01 13:23 ` Kővágó, Zoltán
2015-06-01 13:23 ` [Qemu-devel] [PATCH 05/12] audio: expose drv_opaque to init_out and init_in Kővágó, Zoltán
` (8 subsequent siblings)
12 siblings, 0 replies; 26+ messages in thread
From: Kővágó, Zoltán @ 2015-06-01 13:23 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
DirectSound should be a superior choice on Windows.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/Makefile.objs | 1 -
audio/audio_int.h | 1 -
audio/winwaveaudio.c | 717 ---------------------------------------------------
configure | 11 +-
4 files changed, 2 insertions(+), 728 deletions(-)
delete mode 100644 audio/winwaveaudio.c
diff --git a/audio/Makefile.objs b/audio/Makefile.objs
index 3c84972..89b61e1 100644
--- a/audio/Makefile.objs
+++ b/audio/Makefile.objs
@@ -5,7 +5,6 @@ common-obj-$(CONFIG_COREAUDIO) += coreaudio.o
common-obj-$(CONFIG_ALSA) += alsaaudio.o
common-obj-$(CONFIG_DSOUND) += dsoundaudio.o
common-obj-$(CONFIG_PA) += paaudio.o
-common-obj-$(CONFIG_WINWAVE) += winwaveaudio.o
common-obj-$(CONFIG_AUDIO_PT_INT) += audio_pt_int.o
common-obj-$(CONFIG_AUDIO_WIN_INT) += audio_win_int.o
common-obj-y += wavcapture.o
diff --git a/audio/audio_int.h b/audio/audio_int.h
index d593879..92a2f3c 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -210,7 +210,6 @@ extern struct audio_driver coreaudio_audio_driver;
extern struct audio_driver dsound_audio_driver;
extern struct audio_driver pa_audio_driver;
extern struct audio_driver spice_audio_driver;
-extern struct audio_driver winwave_audio_driver;
extern const struct mixeng_volume nominal_volume;
void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as);
diff --git a/audio/winwaveaudio.c b/audio/winwaveaudio.c
deleted file mode 100644
index 8dbd145..0000000
--- a/audio/winwaveaudio.c
+++ /dev/null
@@ -1,717 +0,0 @@
-/* public domain */
-
-#include "qemu-common.h"
-#include "sysemu/sysemu.h"
-#include "audio.h"
-
-#define AUDIO_CAP "winwave"
-#include "audio_int.h"
-
-#include <windows.h>
-#include <mmsystem.h>
-
-#include "audio_win_int.h"
-
-static struct {
- int dac_headers;
- int dac_samples;
- int adc_headers;
- int adc_samples;
-} conf = {
- .dac_headers = 4,
- .dac_samples = 1024,
- .adc_headers = 4,
- .adc_samples = 1024
-};
-
-typedef struct {
- HWVoiceOut hw;
- HWAVEOUT hwo;
- WAVEHDR *hdrs;
- HANDLE event;
- void *pcm_buf;
- int avail;
- int pending;
- int curhdr;
- int paused;
- CRITICAL_SECTION crit_sect;
-} WaveVoiceOut;
-
-typedef struct {
- HWVoiceIn hw;
- HWAVEIN hwi;
- WAVEHDR *hdrs;
- HANDLE event;
- void *pcm_buf;
- int curhdr;
- int paused;
- int rpos;
- int avail;
- CRITICAL_SECTION crit_sect;
-} WaveVoiceIn;
-
-static void winwave_log_mmresult (MMRESULT mr)
-{
- const char *str = "BUG";
-
- switch (mr) {
- case MMSYSERR_NOERROR:
- str = "Success";
- break;
-
- case MMSYSERR_INVALHANDLE:
- str = "Specified device handle is invalid";
- break;
-
- case MMSYSERR_BADDEVICEID:
- str = "Specified device id is out of range";
- break;
-
- case MMSYSERR_NODRIVER:
- str = "No device driver is present";
- break;
-
- case MMSYSERR_NOMEM:
- str = "Unable to allocate or lock memory";
- break;
-
- case WAVERR_SYNC:
- str = "Device is synchronous but waveOutOpen was called "
- "without using the WINWAVE_ALLOWSYNC flag";
- break;
-
- case WAVERR_UNPREPARED:
- str = "The data block pointed to by the pwh parameter "
- "hasn't been prepared";
- break;
-
- case WAVERR_STILLPLAYING:
- str = "There are still buffers in the queue";
- break;
-
- default:
- dolog ("Reason: Unknown (MMRESULT %#x)\n", mr);
- return;
- }
-
- dolog ("Reason: %s\n", str);
-}
-
-static void GCC_FMT_ATTR (2, 3) winwave_logerr (
- MMRESULT mr,
- const char *fmt,
- ...
- )
-{
- va_list ap;
-
- va_start (ap, fmt);
- AUD_vlog (AUDIO_CAP, fmt, ap);
- va_end (ap);
-
- AUD_log (NULL, " failed\n");
- winwave_log_mmresult (mr);
-}
-
-static void winwave_anal_close_out (WaveVoiceOut *wave)
-{
- MMRESULT mr;
-
- mr = waveOutClose (wave->hwo);
- if (mr != MMSYSERR_NOERROR) {
- winwave_logerr (mr, "waveOutClose");
- }
- wave->hwo = NULL;
-}
-
-static void CALLBACK winwave_callback_out (
- HWAVEOUT hwo,
- UINT msg,
- DWORD_PTR dwInstance,
- DWORD_PTR dwParam1,
- DWORD_PTR dwParam2
- )
-{
- WaveVoiceOut *wave = (WaveVoiceOut *) dwInstance;
-
- switch (msg) {
- case WOM_DONE:
- {
- WAVEHDR *h = (WAVEHDR *) dwParam1;
- if (!h->dwUser) {
- h->dwUser = 1;
- EnterCriticalSection (&wave->crit_sect);
- {
- wave->avail += conf.dac_samples;
- }
- LeaveCriticalSection (&wave->crit_sect);
- if (wave->hw.poll_mode) {
- if (!SetEvent (wave->event)) {
- dolog ("DAC SetEvent failed %lx\n", GetLastError ());
- }
- }
- }
- }
- break;
-
- case WOM_CLOSE:
- case WOM_OPEN:
- break;
-
- default:
- dolog ("unknown wave out callback msg %x\n", msg);
- }
-}
-
-static int winwave_init_out (HWVoiceOut *hw, struct audsettings *as)
-{
- int i;
- int err;
- MMRESULT mr;
- WAVEFORMATEX wfx;
- WaveVoiceOut *wave;
-
- wave = (WaveVoiceOut *) hw;
-
- InitializeCriticalSection (&wave->crit_sect);
-
- err = waveformat_from_audio_settings (&wfx, as);
- if (err) {
- goto err0;
- }
-
- mr = waveOutOpen (&wave->hwo, WAVE_MAPPER, &wfx,
- (DWORD_PTR) winwave_callback_out,
- (DWORD_PTR) wave, CALLBACK_FUNCTION);
- if (mr != MMSYSERR_NOERROR) {
- winwave_logerr (mr, "waveOutOpen");
- goto err1;
- }
-
- wave->hdrs = audio_calloc (AUDIO_FUNC, conf.dac_headers,
- sizeof (*wave->hdrs));
- if (!wave->hdrs) {
- goto err2;
- }
-
- audio_pcm_init_info (&hw->info, as);
- hw->samples = conf.dac_samples * conf.dac_headers;
- wave->avail = hw->samples;
-
- wave->pcm_buf = audio_calloc (AUDIO_FUNC, conf.dac_samples,
- conf.dac_headers << hw->info.shift);
- if (!wave->pcm_buf) {
- goto err3;
- }
-
- for (i = 0; i < conf.dac_headers; ++i) {
- WAVEHDR *h = &wave->hdrs[i];
-
- h->dwUser = 0;
- h->dwBufferLength = conf.dac_samples << hw->info.shift;
- h->lpData = advance (wave->pcm_buf, i * h->dwBufferLength);
- h->dwFlags = 0;
-
- mr = waveOutPrepareHeader (wave->hwo, h, sizeof (*h));
- if (mr != MMSYSERR_NOERROR) {
- winwave_logerr (mr, "waveOutPrepareHeader(%d)", i);
- goto err4;
- }
- }
-
- return 0;
-
- err4:
- g_free (wave->pcm_buf);
- err3:
- g_free (wave->hdrs);
- err2:
- winwave_anal_close_out (wave);
- err1:
- err0:
- return -1;
-}
-
-static int winwave_write (SWVoiceOut *sw, void *buf, int len)
-{
- return audio_pcm_sw_write (sw, buf, len);
-}
-
-static int winwave_run_out (HWVoiceOut *hw, int live)
-{
- WaveVoiceOut *wave = (WaveVoiceOut *) hw;
- int decr;
- int doreset;
-
- EnterCriticalSection (&wave->crit_sect);
- {
- decr = audio_MIN (live, wave->avail);
- decr = audio_pcm_hw_clip_out (hw, wave->pcm_buf, decr, wave->pending);
- wave->pending += decr;
- wave->avail -= decr;
- }
- LeaveCriticalSection (&wave->crit_sect);
-
- doreset = hw->poll_mode && (wave->pending >= conf.dac_samples);
- if (doreset && !ResetEvent (wave->event)) {
- dolog ("DAC ResetEvent failed %lx\n", GetLastError ());
- }
-
- while (wave->pending >= conf.dac_samples) {
- MMRESULT mr;
- WAVEHDR *h = &wave->hdrs[wave->curhdr];
-
- h->dwUser = 0;
- mr = waveOutWrite (wave->hwo, h, sizeof (*h));
- if (mr != MMSYSERR_NOERROR) {
- winwave_logerr (mr, "waveOutWrite(%d)", wave->curhdr);
- break;
- }
-
- wave->pending -= conf.dac_samples;
- wave->curhdr = (wave->curhdr + 1) % conf.dac_headers;
- }
-
- return decr;
-}
-
-static void winwave_poll (void *opaque)
-{
- (void) opaque;
- audio_run ("winwave_poll");
-}
-
-static void winwave_fini_out (HWVoiceOut *hw)
-{
- int i;
- MMRESULT mr;
- WaveVoiceOut *wave = (WaveVoiceOut *) hw;
-
- mr = waveOutReset (wave->hwo);
- if (mr != MMSYSERR_NOERROR) {
- winwave_logerr (mr, "waveOutReset");
- }
-
- for (i = 0; i < conf.dac_headers; ++i) {
- mr = waveOutUnprepareHeader (wave->hwo, &wave->hdrs[i],
- sizeof (wave->hdrs[i]));
- if (mr != MMSYSERR_NOERROR) {
- winwave_logerr (mr, "waveOutUnprepareHeader(%d)", i);
- }
- }
-
- winwave_anal_close_out (wave);
-
- if (wave->event) {
- qemu_del_wait_object (wave->event, winwave_poll, wave);
- if (!CloseHandle (wave->event)) {
- dolog ("DAC CloseHandle failed %lx\n", GetLastError ());
- }
- wave->event = NULL;
- }
-
- g_free (wave->pcm_buf);
- wave->pcm_buf = NULL;
-
- g_free (wave->hdrs);
- wave->hdrs = NULL;
-}
-
-static int winwave_ctl_out (HWVoiceOut *hw, int cmd, ...)
-{
- MMRESULT mr;
- WaveVoiceOut *wave = (WaveVoiceOut *) hw;
-
- switch (cmd) {
- case VOICE_ENABLE:
- {
- va_list ap;
- int poll_mode;
-
- va_start (ap, cmd);
- poll_mode = va_arg (ap, int);
- va_end (ap);
-
- if (poll_mode && !wave->event) {
- wave->event = CreateEvent (NULL, TRUE, TRUE, NULL);
- if (!wave->event) {
- dolog ("DAC CreateEvent: %lx, poll mode will be disabled\n",
- GetLastError ());
- }
- }
-
- if (wave->event) {
- int ret;
-
- ret = qemu_add_wait_object (wave->event, winwave_poll, wave);
- hw->poll_mode = (ret == 0);
- }
- else {
- hw->poll_mode = 0;
- }
- wave->paused = 0;
- }
- return 0;
-
- case VOICE_DISABLE:
- if (!wave->paused) {
- mr = waveOutReset (wave->hwo);
- if (mr != MMSYSERR_NOERROR) {
- winwave_logerr (mr, "waveOutReset");
- }
- else {
- wave->paused = 1;
- }
- }
- if (wave->event) {
- qemu_del_wait_object (wave->event, winwave_poll, wave);
- }
- return 0;
- }
- return -1;
-}
-
-static void winwave_anal_close_in (WaveVoiceIn *wave)
-{
- MMRESULT mr;
-
- mr = waveInClose (wave->hwi);
- if (mr != MMSYSERR_NOERROR) {
- winwave_logerr (mr, "waveInClose");
- }
- wave->hwi = NULL;
-}
-
-static void CALLBACK winwave_callback_in (
- HWAVEIN *hwi,
- UINT msg,
- DWORD_PTR dwInstance,
- DWORD_PTR dwParam1,
- DWORD_PTR dwParam2
- )
-{
- WaveVoiceIn *wave = (WaveVoiceIn *) dwInstance;
-
- switch (msg) {
- case WIM_DATA:
- {
- WAVEHDR *h = (WAVEHDR *) dwParam1;
- if (!h->dwUser) {
- h->dwUser = 1;
- EnterCriticalSection (&wave->crit_sect);
- {
- wave->avail += conf.adc_samples;
- }
- LeaveCriticalSection (&wave->crit_sect);
- if (wave->hw.poll_mode) {
- if (!SetEvent (wave->event)) {
- dolog ("ADC SetEvent failed %lx\n", GetLastError ());
- }
- }
- }
- }
- break;
-
- case WIM_CLOSE:
- case WIM_OPEN:
- break;
-
- default:
- dolog ("unknown wave in callback msg %x\n", msg);
- }
-}
-
-static void winwave_add_buffers (WaveVoiceIn *wave, int samples)
-{
- int doreset;
-
- doreset = wave->hw.poll_mode && (samples >= conf.adc_samples);
- if (doreset && !ResetEvent (wave->event)) {
- dolog ("ADC ResetEvent failed %lx\n", GetLastError ());
- }
-
- while (samples >= conf.adc_samples) {
- MMRESULT mr;
- WAVEHDR *h = &wave->hdrs[wave->curhdr];
-
- h->dwUser = 0;
- mr = waveInAddBuffer (wave->hwi, h, sizeof (*h));
- if (mr != MMSYSERR_NOERROR) {
- winwave_logerr (mr, "waveInAddBuffer(%d)", wave->curhdr);
- }
- wave->curhdr = (wave->curhdr + 1) % conf.adc_headers;
- samples -= conf.adc_samples;
- }
-}
-
-static int winwave_init_in (HWVoiceIn *hw, struct audsettings *as)
-{
- int i;
- int err;
- MMRESULT mr;
- WAVEFORMATEX wfx;
- WaveVoiceIn *wave;
-
- wave = (WaveVoiceIn *) hw;
-
- InitializeCriticalSection (&wave->crit_sect);
-
- err = waveformat_from_audio_settings (&wfx, as);
- if (err) {
- goto err0;
- }
-
- mr = waveInOpen (&wave->hwi, WAVE_MAPPER, &wfx,
- (DWORD_PTR) winwave_callback_in,
- (DWORD_PTR) wave, CALLBACK_FUNCTION);
- if (mr != MMSYSERR_NOERROR) {
- winwave_logerr (mr, "waveInOpen");
- goto err1;
- }
-
- wave->hdrs = audio_calloc (AUDIO_FUNC, conf.dac_headers,
- sizeof (*wave->hdrs));
- if (!wave->hdrs) {
- goto err2;
- }
-
- audio_pcm_init_info (&hw->info, as);
- hw->samples = conf.adc_samples * conf.adc_headers;
- wave->avail = 0;
-
- wave->pcm_buf = audio_calloc (AUDIO_FUNC, conf.adc_samples,
- conf.adc_headers << hw->info.shift);
- if (!wave->pcm_buf) {
- goto err3;
- }
-
- for (i = 0; i < conf.adc_headers; ++i) {
- WAVEHDR *h = &wave->hdrs[i];
-
- h->dwUser = 0;
- h->dwBufferLength = conf.adc_samples << hw->info.shift;
- h->lpData = advance (wave->pcm_buf, i * h->dwBufferLength);
- h->dwFlags = 0;
-
- mr = waveInPrepareHeader (wave->hwi, h, sizeof (*h));
- if (mr != MMSYSERR_NOERROR) {
- winwave_logerr (mr, "waveInPrepareHeader(%d)", i);
- goto err4;
- }
- }
-
- wave->paused = 1;
- winwave_add_buffers (wave, hw->samples);
- return 0;
-
- err4:
- g_free (wave->pcm_buf);
- err3:
- g_free (wave->hdrs);
- err2:
- winwave_anal_close_in (wave);
- err1:
- err0:
- return -1;
-}
-
-static void winwave_fini_in (HWVoiceIn *hw)
-{
- int i;
- MMRESULT mr;
- WaveVoiceIn *wave = (WaveVoiceIn *) hw;
-
- mr = waveInReset (wave->hwi);
- if (mr != MMSYSERR_NOERROR) {
- winwave_logerr (mr, "waveInReset");
- }
-
- for (i = 0; i < conf.adc_headers; ++i) {
- mr = waveInUnprepareHeader (wave->hwi, &wave->hdrs[i],
- sizeof (wave->hdrs[i]));
- if (mr != MMSYSERR_NOERROR) {
- winwave_logerr (mr, "waveInUnprepareHeader(%d)", i);
- }
- }
-
- winwave_anal_close_in (wave);
-
- if (wave->event) {
- qemu_del_wait_object (wave->event, winwave_poll, wave);
- if (!CloseHandle (wave->event)) {
- dolog ("ADC CloseHandle failed %lx\n", GetLastError ());
- }
- wave->event = NULL;
- }
-
- g_free (wave->pcm_buf);
- wave->pcm_buf = NULL;
-
- g_free (wave->hdrs);
- wave->hdrs = NULL;
-}
-
-static int winwave_run_in (HWVoiceIn *hw)
-{
- WaveVoiceIn *wave = (WaveVoiceIn *) hw;
- int live = audio_pcm_hw_get_live_in (hw);
- int dead = hw->samples - live;
- int decr, ret;
-
- if (!dead) {
- return 0;
- }
-
- EnterCriticalSection (&wave->crit_sect);
- {
- decr = audio_MIN (dead, wave->avail);
- wave->avail -= decr;
- }
- LeaveCriticalSection (&wave->crit_sect);
-
- ret = decr;
- while (decr) {
- int left = hw->samples - hw->wpos;
- int conv = audio_MIN (left, decr);
- hw->conv (hw->conv_buf + hw->wpos,
- advance (wave->pcm_buf, wave->rpos << hw->info.shift),
- conv);
-
- wave->rpos = (wave->rpos + conv) % hw->samples;
- hw->wpos = (hw->wpos + conv) % hw->samples;
- decr -= conv;
- }
-
- winwave_add_buffers (wave, ret);
- return ret;
-}
-
-static int winwave_read (SWVoiceIn *sw, void *buf, int size)
-{
- return audio_pcm_sw_read (sw, buf, size);
-}
-
-static int winwave_ctl_in (HWVoiceIn *hw, int cmd, ...)
-{
- MMRESULT mr;
- WaveVoiceIn *wave = (WaveVoiceIn *) hw;
-
- switch (cmd) {
- case VOICE_ENABLE:
- {
- va_list ap;
- int poll_mode;
-
- va_start (ap, cmd);
- poll_mode = va_arg (ap, int);
- va_end (ap);
-
- if (poll_mode && !wave->event) {
- wave->event = CreateEvent (NULL, TRUE, TRUE, NULL);
- if (!wave->event) {
- dolog ("ADC CreateEvent: %lx, poll mode will be disabled\n",
- GetLastError ());
- }
- }
-
- if (wave->event) {
- int ret;
-
- ret = qemu_add_wait_object (wave->event, winwave_poll, wave);
- hw->poll_mode = (ret == 0);
- }
- else {
- hw->poll_mode = 0;
- }
- if (wave->paused) {
- mr = waveInStart (wave->hwi);
- if (mr != MMSYSERR_NOERROR) {
- winwave_logerr (mr, "waveInStart");
- }
- wave->paused = 0;
- }
- }
- return 0;
-
- case VOICE_DISABLE:
- if (!wave->paused) {
- mr = waveInStop (wave->hwi);
- if (mr != MMSYSERR_NOERROR) {
- winwave_logerr (mr, "waveInStop");
- }
- else {
- wave->paused = 1;
- }
- }
- if (wave->event) {
- qemu_del_wait_object (wave->event, winwave_poll, wave);
- }
- return 0;
- }
- return 0;
-}
-
-static void *winwave_audio_init (void)
-{
- return &conf;
-}
-
-static void winwave_audio_fini (void *opaque)
-{
- (void) opaque;
-}
-
-static struct audio_option winwave_options[] = {
- {
- .name = "DAC_HEADERS",
- .tag = AUD_OPT_INT,
- .valp = &conf.dac_headers,
- .descr = "DAC number of headers",
- },
- {
- .name = "DAC_SAMPLES",
- .tag = AUD_OPT_INT,
- .valp = &conf.dac_samples,
- .descr = "DAC number of samples per header",
- },
- {
- .name = "ADC_HEADERS",
- .tag = AUD_OPT_INT,
- .valp = &conf.adc_headers,
- .descr = "ADC number of headers",
- },
- {
- .name = "ADC_SAMPLES",
- .tag = AUD_OPT_INT,
- .valp = &conf.adc_samples,
- .descr = "ADC number of samples per header",
- },
- { /* End of list */ }
-};
-
-static struct audio_pcm_ops winwave_pcm_ops = {
- .init_out = winwave_init_out,
- .fini_out = winwave_fini_out,
- .run_out = winwave_run_out,
- .write = winwave_write,
- .ctl_out = winwave_ctl_out,
- .init_in = winwave_init_in,
- .fini_in = winwave_fini_in,
- .run_in = winwave_run_in,
- .read = winwave_read,
- .ctl_in = winwave_ctl_in
-};
-
-struct audio_driver winwave_audio_driver = {
- .name = "winwave",
- .descr = "Windows Waveform Audio http://msdn.microsoft.com",
- .options = winwave_options,
- .init = winwave_audio_init,
- .fini = winwave_audio_fini,
- .pcm_ops = &winwave_pcm_ops,
- .can_be_default = 1,
- .max_voices_out = INT_MAX,
- .max_voices_in = INT_MAX,
- .voice_size_out = sizeof (WaveVoiceOut),
- .voice_size_in = sizeof (WaveVoiceIn)
-};
diff --git a/configure b/configure
index 154ecbe..b2eddc1 100755
--- a/configure
+++ b/configure
@@ -562,13 +562,11 @@ case $targetos in
CYGWIN*)
mingw32="yes"
QEMU_CFLAGS="-mno-cygwin $QEMU_CFLAGS"
- audio_possible_drivers="winwave"
- audio_drv_list="winwave"
;;
MINGW32*)
mingw32="yes"
- audio_possible_drivers="winwave dsound"
- audio_drv_list="winwave"
+ audio_possible_drivers="dsound"
+ audio_drv_list="dsound"
;;
GNU/kFreeBSD)
bsd="yes"
@@ -2632,11 +2630,6 @@ for drv in $audio_drv_list; do
# XXX: Probes for CoreAudio, DirectSound
;;
- winwave)
- libs_softmmu="-lwinmm $libs_softmmu"
- audio_win_int="yes"
- ;;
-
*)
echo "$audio_possible_drivers" | grep -q "\<$drv\>" || {
error_exit "Unknown driver '$drv' selected" \
--
2.4.2
^ permalink raw reply related [flat|nested] 26+ messages in thread* [Qemu-devel] [PATCH 05/12] audio: expose drv_opaque to init_out and init_in
2015-06-01 13:23 [Qemu-devel] [PATCH 00/12] Audio backend cleanup Kővágó, Zoltán
` (3 preceding siblings ...)
2015-06-01 13:23 ` [Qemu-devel] [PATCH 04/12] audio: remove winwave audio driver Kővágó, Zoltán
@ 2015-06-01 13:23 ` Kővágó, Zoltán
2015-06-02 7:56 ` Gerd Hoffmann
2015-06-01 13:23 ` [Qemu-devel] [PATCH 06/12] alsaaudio: do not use global variables Kővágó, Zoltán
` (7 subsequent siblings)
12 siblings, 1 reply; 26+ messages in thread
From: Kővágó, Zoltán @ 2015-06-01 13:23 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Currently the opaque pointer returned by audio_driver's init is only
exposed to the driver's fini, but not to audio_pcm_ops. This way if
someone wants to share a variable with the driver and the pcm, he must
use global variables. This patch fixes it by adding a third parameter to
audio_pcm_op's init_out and init_in.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/alsaaudio.c | 9 +++++++--
audio/audio_int.h | 4 ++--
audio/audio_template.h | 2 +-
audio/coreaudio.c | 5 ++++-
audio/dsound_template.h | 6 ++++--
audio/noaudio.c | 6 ++++--
audio/ossaudio.c | 9 +++++++--
audio/paaudio.c | 9 +++++++--
audio/spiceaudio.c | 9 +++++++--
audio/wavaudio.c | 4 +++-
10 files changed, 46 insertions(+), 17 deletions(-)
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index 74ead97..eb33bff 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -819,7 +819,8 @@ static void alsa_fini_out (HWVoiceOut *hw)
alsa->pcm_buf = NULL;
}
-static int alsa_init_out (HWVoiceOut *hw, struct audsettings *as)
+static int alsa_init_out(HWVoiceOut *hw, struct audsettings *as,
+ void *drv_opaque)
{
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
struct alsa_params_req req;
@@ -827,6 +828,8 @@ static int alsa_init_out (HWVoiceOut *hw, struct audsettings *as)
snd_pcm_t *handle;
struct audsettings obt_as;
+ (void) drv_opaque;
+
req.fmt = aud_to_alsafmt (as->fmt, as->endianness);
req.freq = as->freq;
req.nchannels = as->nchannels;
@@ -928,7 +931,7 @@ static int alsa_ctl_out (HWVoiceOut *hw, int cmd, ...)
return -1;
}
-static int alsa_init_in (HWVoiceIn *hw, struct audsettings *as)
+static int alsa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
{
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
struct alsa_params_req req;
@@ -936,6 +939,8 @@ static int alsa_init_in (HWVoiceIn *hw, struct audsettings *as)
snd_pcm_t *handle;
struct audsettings obt_as;
+ (void) drv_opaque;
+
req.fmt = aud_to_alsafmt (as->fmt, as->endianness);
req.freq = as->freq;
req.nchannels = as->nchannels;
diff --git a/audio/audio_int.h b/audio/audio_int.h
index 92a2f3c..a3efe7b 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -156,13 +156,13 @@ struct audio_driver {
};
struct audio_pcm_ops {
- int (*init_out)(HWVoiceOut *hw, struct audsettings *as);
+ int (*init_out)(HWVoiceOut *hw, struct audsettings *as, void *drv_opaque);
void (*fini_out)(HWVoiceOut *hw);
int (*run_out) (HWVoiceOut *hw, int live);
int (*write) (SWVoiceOut *sw, void *buf, int size);
int (*ctl_out) (HWVoiceOut *hw, int cmd, ...);
- int (*init_in) (HWVoiceIn *hw, struct audsettings *as);
+ int (*init_in) (HWVoiceIn *hw, struct audsettings *as, void *drv_opaque);
void (*fini_in) (HWVoiceIn *hw);
int (*run_in) (HWVoiceIn *hw);
int (*read) (SWVoiceIn *sw, void *buf, int size);
diff --git a/audio/audio_template.h b/audio/audio_template.h
index 584e536..f716d97 100644
--- a/audio/audio_template.h
+++ b/audio/audio_template.h
@@ -262,7 +262,7 @@ static HW *glue (audio_pcm_hw_add_new_, TYPE) (struct audsettings *as)
#ifdef DAC
QLIST_INIT (&hw->cap_head);
#endif
- if (glue (hw->pcm_ops->init_, TYPE) (hw, as)) {
+ if (glue (hw->pcm_ops->init_, TYPE) (hw, as, s->drv_opaque)) {
goto err0;
}
diff --git a/audio/coreaudio.c b/audio/coreaudio.c
index 5964c62..f38fd82 100644
--- a/audio/coreaudio.c
+++ b/audio/coreaudio.c
@@ -287,7 +287,8 @@ static int coreaudio_write (SWVoiceOut *sw, void *buf, int len)
return audio_pcm_sw_write (sw, buf, len);
}
-static int coreaudio_init_out (HWVoiceOut *hw, struct audsettings *as)
+static int coreaudio_init_out(HWVoiceOut *hw, struct audsettings *as,
+ void *drv_opaque)
{
OSStatus status;
coreaudioVoiceOut *core = (coreaudioVoiceOut *) hw;
@@ -296,6 +297,8 @@ static int coreaudio_init_out (HWVoiceOut *hw, struct audsettings *as)
const char *typ = "playback";
AudioValueRange frameRange;
+ (void) drv_opaque;
+
/* create mutex */
err = pthread_mutex_init(&core->mutex, NULL);
if (err) {
diff --git a/audio/dsound_template.h b/audio/dsound_template.h
index 8b37d16..98276fb 100644
--- a/audio/dsound_template.h
+++ b/audio/dsound_template.h
@@ -174,9 +174,11 @@ static void dsound_fini_out (HWVoiceOut *hw)
}
#ifdef DSBTYPE_IN
-static int dsound_init_in (HWVoiceIn *hw, struct audsettings *as)
+static int dsound_init_in(HWVoiceIn *hw, struct audsettings *as,
+ void *drv_opaque)
#else
-static int dsound_init_out (HWVoiceOut *hw, struct audsettings *as)
+static int dsound_init_out(HWVoiceOut *hw, struct audsettings *as,
+ void *drv_opaque)
#endif
{
int err;
diff --git a/audio/noaudio.c b/audio/noaudio.c
index cb38662..dd1a87a 100644
--- a/audio/noaudio.c
+++ b/audio/noaudio.c
@@ -63,8 +63,9 @@ static int no_write (SWVoiceOut *sw, void *buf, int len)
return audio_pcm_sw_write (sw, buf, len);
}
-static int no_init_out (HWVoiceOut *hw, struct audsettings *as)
+static int no_init_out(HWVoiceOut *hw, struct audsettings *as, void *drv_opaque)
{
+ (void) drv_opaque;
audio_pcm_init_info (&hw->info, as);
hw->samples = 1024;
return 0;
@@ -82,8 +83,9 @@ static int no_ctl_out (HWVoiceOut *hw, int cmd, ...)
return 0;
}
-static int no_init_in (HWVoiceIn *hw, struct audsettings *as)
+static int no_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
{
+ (void) drv_opaque;
audio_pcm_init_info (&hw->info, as);
hw->samples = 1024;
return 0;
diff --git a/audio/ossaudio.c b/audio/ossaudio.c
index 4db2ca6..80ac610 100644
--- a/audio/ossaudio.c
+++ b/audio/ossaudio.c
@@ -510,7 +510,8 @@ static void oss_fini_out (HWVoiceOut *hw)
}
}
-static int oss_init_out (HWVoiceOut *hw, struct audsettings *as)
+static int oss_init_out(HWVoiceOut *hw, struct audsettings *as,
+ void *drv_opaque)
{
OSSVoiceOut *oss = (OSSVoiceOut *) hw;
struct oss_params req, obt;
@@ -520,6 +521,8 @@ static int oss_init_out (HWVoiceOut *hw, struct audsettings *as)
audfmt_e effective_fmt;
struct audsettings obt_as;
+ (void) drv_opaque;
+
oss->fd = -1;
req.fmt = aud_to_ossfmt (as->fmt, as->endianness);
@@ -676,7 +679,7 @@ static int oss_ctl_out (HWVoiceOut *hw, int cmd, ...)
return 0;
}
-static int oss_init_in (HWVoiceIn *hw, struct audsettings *as)
+static int oss_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
{
OSSVoiceIn *oss = (OSSVoiceIn *) hw;
struct oss_params req, obt;
@@ -686,6 +689,8 @@ static int oss_init_in (HWVoiceIn *hw, struct audsettings *as)
audfmt_e effective_fmt;
struct audsettings obt_as;
+ (void) drv_opaque;
+
oss->fd = -1;
req.fmt = aud_to_ossfmt (as->fmt, as->endianness);
diff --git a/audio/paaudio.c b/audio/paaudio.c
index 90ff245..11c2d29 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -534,7 +534,8 @@ fail:
return NULL;
}
-static int qpa_init_out (HWVoiceOut *hw, struct audsettings *as)
+static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as,
+ void *drv_opaque)
{
int error;
static pa_sample_spec ss;
@@ -542,6 +543,8 @@ static int qpa_init_out (HWVoiceOut *hw, struct audsettings *as)
struct audsettings obt_as = *as;
PAVoiceOut *pa = (PAVoiceOut *) hw;
+ (void) drv_opaque;
+
ss.format = audfmt_to_pa (as->fmt, as->endianness);
ss.channels = as->nchannels;
ss.rate = as->freq;
@@ -601,13 +604,15 @@ static int qpa_init_out (HWVoiceOut *hw, struct audsettings *as)
return -1;
}
-static int qpa_init_in (HWVoiceIn *hw, struct audsettings *as)
+static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
{
int error;
static pa_sample_spec ss;
struct audsettings obt_as = *as;
PAVoiceIn *pa = (PAVoiceIn *) hw;
+ (void) drv_opaque;
+
ss.format = audfmt_to_pa (as->fmt, as->endianness);
ss.channels = as->nchannels;
ss.rate = as->freq;
diff --git a/audio/spiceaudio.c b/audio/spiceaudio.c
index 7b79bed..78ca04c 100644
--- a/audio/spiceaudio.c
+++ b/audio/spiceaudio.c
@@ -115,11 +115,14 @@ static int rate_get_samples (struct audio_pcm_info *info, SpiceRateCtl *rate)
/* playback */
-static int line_out_init (HWVoiceOut *hw, struct audsettings *as)
+static int line_out_init(HWVoiceOut *hw, struct audsettings *as,
+ void *drv_opaque)
{
SpiceVoiceOut *out = container_of (hw, SpiceVoiceOut, hw);
struct audsettings settings;
+ (void) drv_opaque;
+
#if SPICE_INTERFACE_PLAYBACK_MAJOR > 1 || SPICE_INTERFACE_PLAYBACK_MINOR >= 3
settings.freq = spice_server_get_best_playback_rate(NULL);
#else
@@ -243,11 +246,13 @@ static int line_out_ctl (HWVoiceOut *hw, int cmd, ...)
/* record */
-static int line_in_init (HWVoiceIn *hw, struct audsettings *as)
+static int line_in_init(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
{
SpiceVoiceIn *in = container_of (hw, SpiceVoiceIn, hw);
struct audsettings settings;
+ (void) drv_opaque;
+
#if SPICE_INTERFACE_RECORD_MAJOR > 2 || SPICE_INTERFACE_RECORD_MINOR >= 3
settings.freq = spice_server_get_best_record_rate(NULL);
#else
diff --git a/audio/wavaudio.c b/audio/wavaudio.c
index 6846a1a..5132aed 100644
--- a/audio/wavaudio.c
+++ b/audio/wavaudio.c
@@ -105,7 +105,8 @@ static void le_store (uint8_t *buf, uint32_t val, int len)
}
}
-static int wav_init_out (HWVoiceOut *hw, struct audsettings *as)
+static int wav_init_out(HWVoiceOut *hw, struct audsettings *as,
+ void *drv_opaque)
{
WAVVoiceOut *wav = (WAVVoiceOut *) hw;
int bits16 = 0, stereo = 0;
@@ -118,6 +119,7 @@ static int wav_init_out (HWVoiceOut *hw, struct audsettings *as)
struct audsettings wav_as = conf.settings;
(void) as;
+ (void) drv_opaque;
stereo = wav_as.nchannels == 2;
switch (wav_as.fmt) {
--
2.4.2
^ permalink raw reply related [flat|nested] 26+ messages in thread* Re: [Qemu-devel] [PATCH 05/12] audio: expose drv_opaque to init_out and init_in
2015-06-01 13:23 ` [Qemu-devel] [PATCH 05/12] audio: expose drv_opaque to init_out and init_in Kővágó, Zoltán
@ 2015-06-02 7:56 ` Gerd Hoffmann
2015-06-02 10:03 ` Kővágó Zoltán
0 siblings, 1 reply; 26+ messages in thread
From: Gerd Hoffmann @ 2015-06-02 7:56 UTC (permalink / raw)
To: Kővágó, Zoltán; +Cc: qemu-devel
Hi,
> -static int alsa_init_out (HWVoiceOut *hw, struct audsettings *as)
> +static int alsa_init_out(HWVoiceOut *hw, struct audsettings *as,
> + void *drv_opaque)
> {
> ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
> struct alsa_params_req req;
> @@ -827,6 +828,8 @@ static int alsa_init_out (HWVoiceOut *hw, struct audsettings *as)
> snd_pcm_t *handle;
> struct audsettings obt_as;
>
> + (void) drv_opaque;
> +
What is the reason for this?
cheers,
Gerd
^ permalink raw reply [flat|nested] 26+ messages in thread* Re: [Qemu-devel] [PATCH 05/12] audio: expose drv_opaque to init_out and init_in
2015-06-02 7:56 ` Gerd Hoffmann
@ 2015-06-02 10:03 ` Kővágó Zoltán
2015-06-02 15:24 ` Gerd Hoffmann
0 siblings, 1 reply; 26+ messages in thread
From: Kővágó Zoltán @ 2015-06-02 10:03 UTC (permalink / raw)
To: Gerd Hoffmann; +Cc: qemu-devel
2015-06-02 09:56 keltezéssel, Gerd Hoffmann írta:
>> -static int alsa_init_out (HWVoiceOut *hw, struct audsettings *as)
>> +static int alsa_init_out(HWVoiceOut *hw, struct audsettings *as,
>> + void *drv_opaque)
>> {
>> ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
>> struct alsa_params_req req;
>> @@ -827,6 +828,8 @@ static int alsa_init_out (HWVoiceOut *hw, struct audsettings *as)
>> snd_pcm_t *handle;
>> struct audsettings obt_as;
>>
>> + (void) drv_opaque;
>> +
>
> What is the reason for this?
To silence compiler warnings about unused variables. I've seen it around
different places in the code. But if they are just historic remains,
I'll remove them.
Thanks,
Zoltan
^ permalink raw reply [flat|nested] 26+ messages in thread* Re: [Qemu-devel] [PATCH 05/12] audio: expose drv_opaque to init_out and init_in
2015-06-02 10:03 ` Kővágó Zoltán
@ 2015-06-02 15:24 ` Gerd Hoffmann
0 siblings, 0 replies; 26+ messages in thread
From: Gerd Hoffmann @ 2015-06-02 15:24 UTC (permalink / raw)
To: Kővágó Zoltán; +Cc: qemu-devel
On Di, 2015-06-02 at 12:03 +0200, Kővágó Zoltán wrote:
> 2015-06-02 09:56 keltezéssel, Gerd Hoffmann írta:
> >> -static int alsa_init_out (HWVoiceOut *hw, struct audsettings *as)
> >> +static int alsa_init_out(HWVoiceOut *hw, struct audsettings *as,
> >> + void *drv_opaque)
> >> {
> >> ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
> >> struct alsa_params_req req;
> >> @@ -827,6 +828,8 @@ static int alsa_init_out (HWVoiceOut *hw, struct audsettings *as)
> >> snd_pcm_t *handle;
> >> struct audsettings obt_as;
> >>
> >> + (void) drv_opaque;
> >> +
> >
> > What is the reason for this?
>
> To silence compiler warnings about unused variables.
That should not happen for parameters (only for local variables).
> I've seen it around
> different places in the code. But if they are just historic remains,
> I'll remove them.
Yes, please.
thanks,
Gerd
^ permalink raw reply [flat|nested] 26+ messages in thread
* [Qemu-devel] [PATCH 06/12] alsaaudio: do not use global variables
2015-06-01 13:23 [Qemu-devel] [PATCH 00/12] Audio backend cleanup Kővágó, Zoltán
` (4 preceding siblings ...)
2015-06-01 13:23 ` [Qemu-devel] [PATCH 05/12] audio: expose drv_opaque to init_out and init_in Kővágó, Zoltán
@ 2015-06-01 13:23 ` Kővágó, Zoltán
2015-06-01 13:23 ` [Qemu-devel] [PATCH 07/12] paaudio: " Kővágó, Zoltán
` (6 subsequent siblings)
12 siblings, 0 replies; 26+ messages in thread
From: Kővágó, Zoltán @ 2015-06-01 13:23 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/alsaaudio.c | 154 +++++++++++++++++++++++++++++-------------------------
1 file changed, 82 insertions(+), 72 deletions(-)
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index eb33bff..d7e181b 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -33,9 +33,29 @@
#define AUDIO_CAP "alsa"
#include "audio_int.h"
+typedef struct ALSAConf {
+ int size_in_usec_in;
+ int size_in_usec_out;
+ const char *pcm_name_in;
+ const char *pcm_name_out;
+ unsigned int buffer_size_in;
+ unsigned int period_size_in;
+ unsigned int buffer_size_out;
+ unsigned int period_size_out;
+ unsigned int threshold;
+
+ int buffer_size_in_overridden;
+ int period_size_in_overridden;
+
+ int buffer_size_out_overridden;
+ int period_size_out_overridden;
+ int verbose;
+} ALSAConf;
+
struct pollhlp {
snd_pcm_t *handle;
struct pollfd *pfds;
+ ALSAConf *conf;
int count;
int mask;
};
@@ -56,30 +76,6 @@ typedef struct ALSAVoiceIn {
struct pollhlp pollhlp;
} ALSAVoiceIn;
-static struct {
- int size_in_usec_in;
- int size_in_usec_out;
- const char *pcm_name_in;
- const char *pcm_name_out;
- unsigned int buffer_size_in;
- unsigned int period_size_in;
- unsigned int buffer_size_out;
- unsigned int period_size_out;
- unsigned int threshold;
-
- int buffer_size_in_overridden;
- int period_size_in_overridden;
-
- int buffer_size_out_overridden;
- int period_size_out_overridden;
- int verbose;
-} conf = {
- .buffer_size_out = 4096,
- .period_size_out = 1024,
- .pcm_name_out = "default",
- .pcm_name_in = "default",
-};
-
struct alsa_params_req {
int freq;
snd_pcm_format_t fmt;
@@ -184,6 +180,7 @@ static void alsa_poll_handler (void *opaque)
snd_pcm_state_t state;
struct pollhlp *hlp = opaque;
unsigned short revents;
+ ALSAConf *conf = hlp->conf;
count = poll (hlp->pfds, hlp->count, 0);
if (count < 0) {
@@ -205,7 +202,7 @@ static void alsa_poll_handler (void *opaque)
}
if (!(revents & hlp->mask)) {
- if (conf.verbose) {
+ if (conf->verbose) {
dolog ("revents = %d\n", revents);
}
return;
@@ -242,6 +239,7 @@ static int alsa_poll_helper (snd_pcm_t *handle, struct pollhlp *hlp, int mask)
{
int i, count, err;
struct pollfd *pfds;
+ ALSAConf *conf = hlp->conf;
count = snd_pcm_poll_descriptors_count (handle);
if (count <= 0) {
@@ -270,13 +268,13 @@ static int alsa_poll_helper (snd_pcm_t *handle, struct pollhlp *hlp, int mask)
NULL, hlp);
}
if (pfds[i].events & POLLOUT) {
- if (conf.verbose) {
+ if (conf->verbose) {
dolog ("POLLOUT %d %d\n", i, pfds[i].fd);
}
err = qemu_set_fd_handler (pfds[i].fd, NULL,
alsa_poll_handler, hlp);
}
- if (conf.verbose) {
+ if (conf->verbose) {
dolog ("Set handler events=%#x index=%d fd=%d err=%d\n",
pfds[i].events, i, pfds[i].fd, err);
}
@@ -476,14 +474,15 @@ static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold)
}
static int alsa_open (int in, struct alsa_params_req *req,
- struct alsa_params_obt *obt, snd_pcm_t **handlep)
+ struct alsa_params_obt *obt, snd_pcm_t **handlep,
+ ALSAConf *conf)
{
snd_pcm_t *handle;
snd_pcm_hw_params_t *hw_params;
int err;
int size_in_usec;
unsigned int freq, nchannels;
- const char *pcm_name = in ? conf.pcm_name_in : conf.pcm_name_out;
+ const char *pcm_name = in ? conf->pcm_name_in : conf->pcm_name_out;
snd_pcm_uframes_t obt_buffer_size;
const char *typ = in ? "ADC" : "DAC";
snd_pcm_format_t obtfmt;
@@ -522,7 +521,7 @@ static int alsa_open (int in, struct alsa_params_req *req,
}
err = snd_pcm_hw_params_set_format (handle, hw_params, req->fmt);
- if (err < 0 && conf.verbose) {
+ if (err < 0 && conf->verbose) {
alsa_logerr2 (err, typ, "Failed to set format %d\n", req->fmt);
}
@@ -654,7 +653,7 @@ static int alsa_open (int in, struct alsa_params_req *req,
goto err;
}
- if (!in && conf.threshold) {
+ if (!in && conf->threshold) {
snd_pcm_uframes_t threshold;
int bytes_per_sec;
@@ -676,7 +675,7 @@ static int alsa_open (int in, struct alsa_params_req *req,
break;
}
- threshold = (conf.threshold * bytes_per_sec) / 1000;
+ threshold = (conf->threshold * bytes_per_sec) / 1000;
alsa_set_threshold (handle, threshold);
}
@@ -686,7 +685,7 @@ static int alsa_open (int in, struct alsa_params_req *req,
*handlep = handle;
- if (conf.verbose &&
+ if (conf->verbose &&
(obtfmt != req->fmt ||
obt->nchannels != req->nchannels ||
obt->freq != req->freq)) {
@@ -729,6 +728,7 @@ static snd_pcm_sframes_t alsa_get_avail (snd_pcm_t *handle)
static void alsa_write_pending (ALSAVoiceOut *alsa)
{
HWVoiceOut *hw = &alsa->hw;
+ ALSAConf *conf = alsa->pollhlp.conf;
while (alsa->pending) {
int left_till_end_samples = hw->samples - alsa->wpos;
@@ -743,7 +743,7 @@ static void alsa_write_pending (ALSAVoiceOut *alsa)
if (written <= 0) {
switch (written) {
case 0:
- if (conf.verbose) {
+ if (conf->verbose) {
dolog ("Failed to write %d frames (wrote zero)\n", len);
}
return;
@@ -754,7 +754,7 @@ static void alsa_write_pending (ALSAVoiceOut *alsa)
len);
return;
}
- if (conf.verbose) {
+ if (conf->verbose) {
dolog ("Recovering from playback xrun\n");
}
continue;
@@ -767,7 +767,7 @@ static void alsa_write_pending (ALSAVoiceOut *alsa)
len);
return;
}
- if (conf.verbose) {
+ if (conf->verbose) {
dolog ("Resuming suspended output stream\n");
}
continue;
@@ -827,20 +827,19 @@ static int alsa_init_out(HWVoiceOut *hw, struct audsettings *as,
struct alsa_params_obt obt;
snd_pcm_t *handle;
struct audsettings obt_as;
-
- (void) drv_opaque;
+ ALSAConf *conf = drv_opaque;
req.fmt = aud_to_alsafmt (as->fmt, as->endianness);
req.freq = as->freq;
req.nchannels = as->nchannels;
- req.period_size = conf.period_size_out;
- req.buffer_size = conf.buffer_size_out;
- req.size_in_usec = conf.size_in_usec_out;
+ req.period_size = conf->period_size_out;
+ req.buffer_size = conf->buffer_size_out;
+ req.size_in_usec = conf->size_in_usec_out;
req.override_mask =
- (conf.period_size_out_overridden ? 1 : 0) |
- (conf.buffer_size_out_overridden ? 2 : 0);
+ (conf->period_size_out_overridden ? 1 : 0) |
+ (conf->buffer_size_out_overridden ? 2 : 0);
- if (alsa_open (0, &req, &obt, &handle)) {
+ if (alsa_open (0, &req, &obt, &handle, conf)) {
return -1;
}
@@ -861,6 +860,7 @@ static int alsa_init_out(HWVoiceOut *hw, struct audsettings *as,
}
alsa->handle = handle;
+ alsa->pollhlp.conf = conf;
return 0;
}
@@ -938,20 +938,19 @@ static int alsa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
struct alsa_params_obt obt;
snd_pcm_t *handle;
struct audsettings obt_as;
-
- (void) drv_opaque;
+ ALSAConf *conf = drv_opaque;
req.fmt = aud_to_alsafmt (as->fmt, as->endianness);
req.freq = as->freq;
req.nchannels = as->nchannels;
- req.period_size = conf.period_size_in;
- req.buffer_size = conf.buffer_size_in;
- req.size_in_usec = conf.size_in_usec_in;
+ req.period_size = conf->period_size_in;
+ req.buffer_size = conf->buffer_size_in;
+ req.size_in_usec = conf->size_in_usec_in;
req.override_mask =
- (conf.period_size_in_overridden ? 1 : 0) |
- (conf.buffer_size_in_overridden ? 2 : 0);
+ (conf->period_size_in_overridden ? 1 : 0) |
+ (conf->buffer_size_in_overridden ? 2 : 0);
- if (alsa_open (1, &req, &obt, &handle)) {
+ if (alsa_open (1, &req, &obt, &handle, conf)) {
return -1;
}
@@ -972,6 +971,7 @@ static int alsa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
}
alsa->handle = handle;
+ alsa->pollhlp.conf = conf;
return 0;
}
@@ -1002,6 +1002,7 @@ static int alsa_run_in (HWVoiceIn *hw)
};
snd_pcm_sframes_t avail;
snd_pcm_uframes_t read_samples = 0;
+ ALSAConf *conf = alsa->pollhlp.conf;
if (!dead) {
return 0;
@@ -1027,12 +1028,12 @@ static int alsa_run_in (HWVoiceIn *hw)
dolog ("Failed to resume suspended input stream\n");
return 0;
}
- if (conf.verbose) {
+ if (conf->verbose) {
dolog ("Resuming suspended input stream\n");
}
break;
default:
- if (conf.verbose) {
+ if (conf->verbose) {
dolog ("No frames available and ALSA state is %d\n", state);
}
return 0;
@@ -1069,7 +1070,7 @@ static int alsa_run_in (HWVoiceIn *hw)
if (nread <= 0) {
switch (nread) {
case 0:
- if (conf.verbose) {
+ if (conf->verbose) {
dolog ("Failed to read %ld frames (read zero)\n", len);
}
goto exit;
@@ -1079,7 +1080,7 @@ static int alsa_run_in (HWVoiceIn *hw)
alsa_logerr (nread, "Failed to read %ld frames\n", len);
goto exit;
}
- if (conf.verbose) {
+ if (conf->verbose) {
dolog ("Recovering from capture xrun\n");
}
continue;
@@ -1153,80 +1154,89 @@ static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...)
return -1;
}
+static ALSAConf glob_conf = {
+ .buffer_size_out = 4096,
+ .period_size_out = 1024,
+ .pcm_name_out = "default",
+ .pcm_name_in = "default",
+};
+
static void *alsa_audio_init (void)
{
- return &conf;
+ ALSAConf *conf = g_malloc(sizeof(ALSAConf));
+ *conf = glob_conf;
+ return conf;
}
static void alsa_audio_fini (void *opaque)
{
- (void) opaque;
+ g_free(opaque);
}
static struct audio_option alsa_options[] = {
{
.name = "DAC_SIZE_IN_USEC",
.tag = AUD_OPT_BOOL,
- .valp = &conf.size_in_usec_out,
+ .valp = &glob_conf.size_in_usec_out,
.descr = "DAC period/buffer size in microseconds (otherwise in frames)"
},
{
.name = "DAC_PERIOD_SIZE",
.tag = AUD_OPT_INT,
- .valp = &conf.period_size_out,
+ .valp = &glob_conf.period_size_out,
.descr = "DAC period size (0 to go with system default)",
- .overriddenp = &conf.period_size_out_overridden
+ .overriddenp = &glob_conf.period_size_out_overridden
},
{
.name = "DAC_BUFFER_SIZE",
.tag = AUD_OPT_INT,
- .valp = &conf.buffer_size_out,
+ .valp = &glob_conf.buffer_size_out,
.descr = "DAC buffer size (0 to go with system default)",
- .overriddenp = &conf.buffer_size_out_overridden
+ .overriddenp = &glob_conf.buffer_size_out_overridden
},
{
.name = "ADC_SIZE_IN_USEC",
.tag = AUD_OPT_BOOL,
- .valp = &conf.size_in_usec_in,
+ .valp = &glob_conf.size_in_usec_in,
.descr =
"ADC period/buffer size in microseconds (otherwise in frames)"
},
{
.name = "ADC_PERIOD_SIZE",
.tag = AUD_OPT_INT,
- .valp = &conf.period_size_in,
+ .valp = &glob_conf.period_size_in,
.descr = "ADC period size (0 to go with system default)",
- .overriddenp = &conf.period_size_in_overridden
+ .overriddenp = &glob_conf.period_size_in_overridden
},
{
.name = "ADC_BUFFER_SIZE",
.tag = AUD_OPT_INT,
- .valp = &conf.buffer_size_in,
+ .valp = &glob_conf.buffer_size_in,
.descr = "ADC buffer size (0 to go with system default)",
- .overriddenp = &conf.buffer_size_in_overridden
+ .overriddenp = &glob_conf.buffer_size_in_overridden
},
{
.name = "THRESHOLD",
.tag = AUD_OPT_INT,
- .valp = &conf.threshold,
+ .valp = &glob_conf.threshold,
.descr = "(undocumented)"
},
{
.name = "DAC_DEV",
.tag = AUD_OPT_STR,
- .valp = &conf.pcm_name_out,
+ .valp = &glob_conf.pcm_name_out,
.descr = "DAC device name (for instance dmix)"
},
{
.name = "ADC_DEV",
.tag = AUD_OPT_STR,
- .valp = &conf.pcm_name_in,
+ .valp = &glob_conf.pcm_name_in,
.descr = "ADC device name"
},
{
.name = "VERBOSE",
.tag = AUD_OPT_BOOL,
- .valp = &conf.verbose,
+ .valp = &glob_conf.verbose,
.descr = "Behave in a more verbose way"
},
{ /* End of list */ }
--
2.4.2
^ permalink raw reply related [flat|nested] 26+ messages in thread* [Qemu-devel] [PATCH 07/12] paaudio: do not use global variables
2015-06-01 13:23 [Qemu-devel] [PATCH 00/12] Audio backend cleanup Kővágó, Zoltán
` (5 preceding siblings ...)
2015-06-01 13:23 ` [Qemu-devel] [PATCH 06/12] alsaaudio: do not use global variables Kővágó, Zoltán
@ 2015-06-01 13:23 ` Kővágó, Zoltán
2015-06-01 13:23 ` [Qemu-devel] [PATCH 08/12] ossaudio: " Kővágó, Zoltán
` (5 subsequent siblings)
12 siblings, 0 replies; 26+ messages in thread
From: Kővágó, Zoltán @ 2015-06-01 13:23 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/paaudio.c | 102 +++++++++++++++++++++++++++++---------------------------
1 file changed, 52 insertions(+), 50 deletions(-)
diff --git a/audio/paaudio.c b/audio/paaudio.c
index 11c2d29..35e8887 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -9,6 +9,19 @@
#include "audio_pt_int.h"
typedef struct {
+ int samples;
+ char *server;
+ char *sink;
+ char *source;
+} PAConf;
+
+typedef struct {
+ PAConf conf;
+ pa_threaded_mainloop *mainloop;
+ pa_context *context;
+} paaudio;
+
+typedef struct {
HWVoiceOut hw;
int done;
int live;
@@ -17,6 +30,7 @@ typedef struct {
pa_stream *stream;
void *pcm_buf;
struct audio_pt pt;
+ paaudio *g;
} PAVoiceOut;
typedef struct {
@@ -30,21 +44,9 @@ typedef struct {
struct audio_pt pt;
const void *read_data;
size_t read_index, read_length;
+ paaudio *g;
} PAVoiceIn;
-typedef struct {
- int samples;
- char *server;
- char *sink;
- char *source;
- pa_threaded_mainloop *mainloop;
- pa_context *context;
-} paaudio;
-
-static paaudio glob_paaudio = {
- .samples = 4096,
-};
-
static void GCC_FMT_ATTR (2, 3) qpa_logerr (int err, const char *fmt, ...)
{
va_list ap;
@@ -106,7 +108,7 @@ static inline int PA_STREAM_IS_GOOD(pa_stream_state_t x)
static int qpa_simple_read (PAVoiceIn *p, void *data, size_t length, int *rerror)
{
- paaudio *g = &glob_paaudio;
+ paaudio *g = p->g;
pa_threaded_mainloop_lock (g->mainloop);
@@ -160,7 +162,7 @@ unlock_and_fail:
static int qpa_simple_write (PAVoiceOut *p, const void *data, size_t length, int *rerror)
{
- paaudio *g = &glob_paaudio;
+ paaudio *g = p->g;
pa_threaded_mainloop_lock (g->mainloop);
@@ -222,7 +224,7 @@ static void *qpa_thread_out (void *arg)
}
}
- decr = to_mix = audio_MIN (pa->live, glob_paaudio.samples >> 2);
+ decr = to_mix = audio_MIN (pa->live, pa->g->conf.samples >> 2);
rpos = pa->rpos;
if (audio_pt_unlock (&pa->pt, AUDIO_FUNC)) {
@@ -314,7 +316,7 @@ static void *qpa_thread_in (void *arg)
}
}
- incr = to_grab = audio_MIN (pa->dead, glob_paaudio.samples >> 2);
+ incr = to_grab = audio_MIN (pa->dead, pa->g->conf.samples >> 2);
wpos = pa->wpos;
if (audio_pt_unlock (&pa->pt, AUDIO_FUNC)) {
@@ -430,7 +432,7 @@ static audfmt_e pa_to_audfmt (pa_sample_format_t fmt, int *endianness)
static void context_state_cb (pa_context *c, void *userdata)
{
- paaudio *g = &glob_paaudio;
+ paaudio *g = userdata;
switch (pa_context_get_state(c)) {
case PA_CONTEXT_READY:
@@ -449,7 +451,7 @@ static void context_state_cb (pa_context *c, void *userdata)
static void stream_state_cb (pa_stream *s, void * userdata)
{
- paaudio *g = &glob_paaudio;
+ paaudio *g = userdata;
switch (pa_stream_get_state (s)) {
@@ -467,23 +469,21 @@ static void stream_state_cb (pa_stream *s, void * userdata)
static void stream_request_cb (pa_stream *s, size_t length, void *userdata)
{
- paaudio *g = &glob_paaudio;
+ paaudio *g = userdata;
pa_threaded_mainloop_signal (g->mainloop, 0);
}
static pa_stream *qpa_simple_new (
- const char *server,
+ paaudio *g,
const char *name,
pa_stream_direction_t dir,
const char *dev,
- const char *stream_name,
const pa_sample_spec *ss,
const pa_channel_map *map,
const pa_buffer_attr *attr,
int *rerror)
{
- paaudio *g = &glob_paaudio;
int r;
pa_stream *stream;
@@ -538,12 +538,11 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as,
void *drv_opaque)
{
int error;
- static pa_sample_spec ss;
- static pa_buffer_attr ba;
+ pa_sample_spec ss;
+ pa_buffer_attr ba;
struct audsettings obt_as = *as;
PAVoiceOut *pa = (PAVoiceOut *) hw;
-
- (void) drv_opaque;
+ paaudio *g = pa->g = drv_opaque;
ss.format = audfmt_to_pa (as->fmt, as->endianness);
ss.channels = as->nchannels;
@@ -561,11 +560,10 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as,
obt_as.fmt = pa_to_audfmt (ss.format, &obt_as.endianness);
pa->stream = qpa_simple_new (
- glob_paaudio.server,
+ g,
"qemu",
PA_STREAM_PLAYBACK,
- glob_paaudio.sink,
- "pcm.playback",
+ g->conf.sink,
&ss,
NULL, /* channel map */
&ba, /* buffering attributes */
@@ -577,7 +575,7 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as,
}
audio_pcm_init_info (&hw->info, &obt_as);
- hw->samples = glob_paaudio.samples;
+ hw->samples = g->conf.samples;
pa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
pa->rpos = hw->rpos;
if (!pa->pcm_buf) {
@@ -607,11 +605,10 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as,
static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
{
int error;
- static pa_sample_spec ss;
+ pa_sample_spec ss;
struct audsettings obt_as = *as;
PAVoiceIn *pa = (PAVoiceIn *) hw;
-
- (void) drv_opaque;
+ paaudio *g = pa->g = drv_opaque;
ss.format = audfmt_to_pa (as->fmt, as->endianness);
ss.channels = as->nchannels;
@@ -620,11 +617,10 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
obt_as.fmt = pa_to_audfmt (ss.format, &obt_as.endianness);
pa->stream = qpa_simple_new (
- glob_paaudio.server,
+ g,
"qemu",
PA_STREAM_RECORD,
- glob_paaudio.source,
- "pcm.capture",
+ g->conf.source,
&ss,
NULL, /* channel map */
NULL, /* buffering attributes */
@@ -636,7 +632,7 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
}
audio_pcm_init_info (&hw->info, &obt_as);
- hw->samples = glob_paaudio.samples;
+ hw->samples = g->conf.samples;
pa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
pa->wpos = hw->wpos;
if (!pa->pcm_buf) {
@@ -708,7 +704,7 @@ static int qpa_ctl_out (HWVoiceOut *hw, int cmd, ...)
PAVoiceOut *pa = (PAVoiceOut *) hw;
pa_operation *op;
pa_cvolume v;
- paaudio *g = &glob_paaudio;
+ paaudio *g = pa->g;
#ifdef PA_CHECK_VERSION /* macro is present in 0.9.16+ */
pa_cvolume_init (&v); /* function is present in 0.9.13+ */
@@ -760,7 +756,7 @@ static int qpa_ctl_in (HWVoiceIn *hw, int cmd, ...)
PAVoiceIn *pa = (PAVoiceIn *) hw;
pa_operation *op;
pa_cvolume v;
- paaudio *g = &glob_paaudio;
+ paaudio *g = pa->g;
#ifdef PA_CHECK_VERSION
pa_cvolume_init (&v);
@@ -810,23 +806,29 @@ static int qpa_ctl_in (HWVoiceIn *hw, int cmd, ...)
}
/* common */
+static PAConf glob_conf = {
+ .samples = 4096,
+};
+
static void *qpa_audio_init (void)
{
- paaudio *g = &glob_paaudio;
+ paaudio *g = g_malloc(sizeof(paaudio));
+ g->conf = glob_conf;
g->mainloop = pa_threaded_mainloop_new ();
if (!g->mainloop) {
goto fail;
}
- g->context = pa_context_new (pa_threaded_mainloop_get_api (g->mainloop), glob_paaudio.server);
+ g->context = pa_context_new (pa_threaded_mainloop_get_api (g->mainloop),
+ g->conf.server);
if (!g->context) {
goto fail;
}
pa_context_set_state_callback (g->context, context_state_cb, g);
- if (pa_context_connect (g->context, glob_paaudio.server, 0, NULL) < 0) {
+ if (pa_context_connect (g->context, g->conf.server, 0, NULL) < 0) {
qpa_logerr (pa_context_errno (g->context),
"pa_context_connect() failed\n");
goto fail;
@@ -859,12 +861,13 @@ static void *qpa_audio_init (void)
pa_threaded_mainloop_unlock (g->mainloop);
- return &glob_paaudio;
+ return g;
unlock_and_fail:
pa_threaded_mainloop_unlock (g->mainloop);
fail:
AUD_log (AUDIO_CAP, "Failed to initialize PA context");
+ g_free(g);
return NULL;
}
@@ -879,39 +882,38 @@ static void qpa_audio_fini (void *opaque)
if (g->context) {
pa_context_disconnect (g->context);
pa_context_unref (g->context);
- g->context = NULL;
}
if (g->mainloop) {
pa_threaded_mainloop_free (g->mainloop);
}
- g->mainloop = NULL;
+ g_free(g);
}
struct audio_option qpa_options[] = {
{
.name = "SAMPLES",
.tag = AUD_OPT_INT,
- .valp = &glob_paaudio.samples,
+ .valp = &glob_conf.samples,
.descr = "buffer size in samples"
},
{
.name = "SERVER",
.tag = AUD_OPT_STR,
- .valp = &glob_paaudio.server,
+ .valp = &glob_conf.server,
.descr = "server address"
},
{
.name = "SINK",
.tag = AUD_OPT_STR,
- .valp = &glob_paaudio.sink,
+ .valp = &glob_conf.sink,
.descr = "sink device name"
},
{
.name = "SOURCE",
.tag = AUD_OPT_STR,
- .valp = &glob_paaudio.source,
+ .valp = &glob_conf.source,
.descr = "source device name"
},
{ /* End of list */ }
--
2.4.2
^ permalink raw reply related [flat|nested] 26+ messages in thread* [Qemu-devel] [PATCH 08/12] ossaudio: do not use global variables
2015-06-01 13:23 [Qemu-devel] [PATCH 00/12] Audio backend cleanup Kővágó, Zoltán
` (6 preceding siblings ...)
2015-06-01 13:23 ` [Qemu-devel] [PATCH 07/12] paaudio: " Kővágó, Zoltán
@ 2015-06-01 13:23 ` Kővágó, Zoltán
2015-06-01 13:23 ` [Qemu-devel] [PATCH 09/12] wavaudio: " Kővágó, Zoltán
` (4 subsequent siblings)
12 siblings, 0 replies; 26+ messages in thread
From: Kővágó, Zoltán @ 2015-06-01 13:23 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/ossaudio.c | 114 +++++++++++++++++++++++++++++--------------------------
1 file changed, 61 insertions(+), 53 deletions(-)
diff --git a/audio/ossaudio.c b/audio/ossaudio.c
index 80ac610..d247969 100644
--- a/audio/ossaudio.c
+++ b/audio/ossaudio.c
@@ -38,6 +38,17 @@
#define USE_DSP_POLICY
#endif
+typedef struct OSSConf {
+ int try_mmap;
+ int nfrags;
+ int fragsize;
+ const char *devpath_out;
+ const char *devpath_in;
+ int debug;
+ int exclusive;
+ int policy;
+} OSSConf;
+
typedef struct OSSVoiceOut {
HWVoiceOut hw;
void *pcm_buf;
@@ -47,6 +58,7 @@ typedef struct OSSVoiceOut {
int fragsize;
int mmapped;
int pending;
+ OSSConf *conf;
} OSSVoiceOut;
typedef struct OSSVoiceIn {
@@ -55,28 +67,9 @@ typedef struct OSSVoiceIn {
int fd;
int nfrags;
int fragsize;
+ OSSConf *conf;
} OSSVoiceIn;
-static struct {
- int try_mmap;
- int nfrags;
- int fragsize;
- const char *devpath_out;
- const char *devpath_in;
- int debug;
- int exclusive;
- int policy;
-} conf = {
- .try_mmap = 0,
- .nfrags = 4,
- .fragsize = 4096,
- .devpath_out = "/dev/dsp",
- .devpath_in = "/dev/dsp",
- .debug = 0,
- .exclusive = 0,
- .policy = 5
-};
-
struct oss_params {
int freq;
audfmt_e fmt;
@@ -272,18 +265,18 @@ static int oss_get_version (int fd, int *version, const char *typ)
#endif
static int oss_open (int in, struct oss_params *req,
- struct oss_params *obt, int *pfd)
+ struct oss_params *obt, int *pfd, OSSConf* conf)
{
int fd;
- int oflags = conf.exclusive ? O_EXCL : 0;
+ int oflags = conf->exclusive ? O_EXCL : 0;
audio_buf_info abinfo;
int fmt, freq, nchannels;
int setfragment = 1;
- const char *dspname = in ? conf.devpath_in : conf.devpath_out;
+ const char *dspname = in ? conf->devpath_in : conf->devpath_out;
const char *typ = in ? "ADC" : "DAC";
/* Kludge needed to have working mmap on Linux */
- oflags |= conf.try_mmap ? O_RDWR : (in ? O_RDONLY : O_WRONLY);
+ oflags |= conf->try_mmap ? O_RDWR : (in ? O_RDONLY : O_WRONLY);
fd = open (dspname, oflags | O_NONBLOCK);
if (-1 == fd) {
@@ -317,20 +310,20 @@ static int oss_open (int in, struct oss_params *req,
}
#ifdef USE_DSP_POLICY
- if (conf.policy >= 0) {
+ if (conf->policy >= 0) {
int version;
if (!oss_get_version (fd, &version, typ)) {
- if (conf.debug) {
+ if (conf->debug) {
dolog ("OSS version = %#x\n", version);
}
if (version >= 0x040000) {
- int policy = conf.policy;
+ int policy = conf->policy;
if (ioctl (fd, SNDCTL_DSP_POLICY, &policy)) {
oss_logerr2 (errno, typ,
"Failed to set timing policy to %d\n",
- conf.policy);
+ conf->policy);
goto err;
}
setfragment = 0;
@@ -434,6 +427,7 @@ static int oss_run_out (HWVoiceOut *hw, int live)
struct audio_buf_info abinfo;
struct count_info cntinfo;
int bufsize;
+ OSSConf *conf = oss->conf;
bufsize = hw->samples << hw->info.shift;
@@ -458,7 +452,7 @@ static int oss_run_out (HWVoiceOut *hw, int live)
}
if (abinfo.bytes > bufsize) {
- if (conf.debug) {
+ if (conf->debug) {
dolog ("warning: Invalid available size, size=%d bufsize=%d\n"
"please report your OS/audio hw to av1474@comtv.ru\n",
abinfo.bytes, bufsize);
@@ -467,7 +461,7 @@ static int oss_run_out (HWVoiceOut *hw, int live)
}
if (abinfo.bytes < 0) {
- if (conf.debug) {
+ if (conf->debug) {
dolog ("warning: Invalid available size, size=%d bufsize=%d\n",
abinfo.bytes, bufsize);
}
@@ -520,18 +514,17 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings *as,
int fd;
audfmt_e effective_fmt;
struct audsettings obt_as;
-
- (void) drv_opaque;
+ OSSConf *conf = drv_opaque;
oss->fd = -1;
req.fmt = aud_to_ossfmt (as->fmt, as->endianness);
req.freq = as->freq;
req.nchannels = as->nchannels;
- req.fragsize = conf.fragsize;
- req.nfrags = conf.nfrags;
+ req.fragsize = conf->fragsize;
+ req.nfrags = conf->nfrags;
- if (oss_open (0, &req, &obt, &fd)) {
+ if (oss_open (0, &req, &obt, &fd, conf)) {
return -1;
}
@@ -558,7 +551,7 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings *as,
hw->samples = (obt.nfrags * obt.fragsize) >> hw->info.shift;
oss->mmapped = 0;
- if (conf.try_mmap) {
+ if (conf->try_mmap) {
oss->pcm_buf = mmap (
NULL,
hw->samples << hw->info.shift,
@@ -618,6 +611,7 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings *as,
}
oss->fd = fd;
+ oss->conf = conf;
return 0;
}
@@ -688,17 +682,16 @@ static int oss_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
int fd;
audfmt_e effective_fmt;
struct audsettings obt_as;
-
- (void) drv_opaque;
+ OSSConf *conf = drv_opaque;
oss->fd = -1;
req.fmt = aud_to_ossfmt (as->fmt, as->endianness);
req.freq = as->freq;
req.nchannels = as->nchannels;
- req.fragsize = conf.fragsize;
- req.nfrags = conf.nfrags;
- if (oss_open (1, &req, &obt, &fd)) {
+ req.fragsize = conf->fragsize;
+ req.nfrags = conf->nfrags;
+ if (oss_open (1, &req, &obt, &fd, conf)) {
return -1;
}
@@ -732,6 +725,7 @@ static int oss_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
}
oss->fd = fd;
+ oss->conf = conf;
return 0;
}
@@ -850,69 +844,83 @@ static int oss_ctl_in (HWVoiceIn *hw, int cmd, ...)
return 0;
}
+static OSSConf glob_conf = {
+ .try_mmap = 0,
+ .nfrags = 4,
+ .fragsize = 4096,
+ .devpath_out = "/dev/dsp",
+ .devpath_in = "/dev/dsp",
+ .debug = 0,
+ .exclusive = 0,
+ .policy = 5
+};
+
static void *oss_audio_init (void)
{
- if (access(conf.devpath_in, R_OK | W_OK) < 0 ||
- access(conf.devpath_out, R_OK | W_OK) < 0) {
+ OSSConf *conf = g_malloc(sizeof(OSSConf));
+ *conf = glob_conf;
+
+ if (access(conf->devpath_in, R_OK | W_OK) < 0 ||
+ access(conf->devpath_out, R_OK | W_OK) < 0) {
return NULL;
}
- return &conf;
+ return conf;
}
static void oss_audio_fini (void *opaque)
{
- (void) opaque;
+ g_free(opaque);
}
static struct audio_option oss_options[] = {
{
.name = "FRAGSIZE",
.tag = AUD_OPT_INT,
- .valp = &conf.fragsize,
+ .valp = &glob_conf.fragsize,
.descr = "Fragment size in bytes"
},
{
.name = "NFRAGS",
.tag = AUD_OPT_INT,
- .valp = &conf.nfrags,
+ .valp = &glob_conf.nfrags,
.descr = "Number of fragments"
},
{
.name = "MMAP",
.tag = AUD_OPT_BOOL,
- .valp = &conf.try_mmap,
+ .valp = &glob_conf.try_mmap,
.descr = "Try using memory mapped access"
},
{
.name = "DAC_DEV",
.tag = AUD_OPT_STR,
- .valp = &conf.devpath_out,
+ .valp = &glob_conf.devpath_out,
.descr = "Path to DAC device"
},
{
.name = "ADC_DEV",
.tag = AUD_OPT_STR,
- .valp = &conf.devpath_in,
+ .valp = &glob_conf.devpath_in,
.descr = "Path to ADC device"
},
{
.name = "EXCLUSIVE",
.tag = AUD_OPT_BOOL,
- .valp = &conf.exclusive,
+ .valp = &glob_conf.exclusive,
.descr = "Open device in exclusive mode (vmix wont work)"
},
#ifdef USE_DSP_POLICY
{
.name = "POLICY",
.tag = AUD_OPT_INT,
- .valp = &conf.policy,
+ .valp = &glob_conf.policy,
.descr = "Set the timing policy of the device, -1 to use fragment mode",
},
#endif
{
.name = "DEBUG",
.tag = AUD_OPT_BOOL,
- .valp = &conf.debug,
+ .valp = &glob_conf.debug,
.descr = "Turn on some debugging messages"
},
{ /* End of list */ }
--
2.4.2
^ permalink raw reply related [flat|nested] 26+ messages in thread* [Qemu-devel] [PATCH 09/12] wavaudio: do not use global variables
2015-06-01 13:23 [Qemu-devel] [PATCH 00/12] Audio backend cleanup Kővágó, Zoltán
` (7 preceding siblings ...)
2015-06-01 13:23 ` [Qemu-devel] [PATCH 08/12] ossaudio: " Kővágó, Zoltán
@ 2015-06-01 13:23 ` Kővágó, Zoltán
2015-06-01 13:23 ` [Qemu-devel] [PATCH 10/12] dsoundaudio: " Kővágó, Zoltán
` (3 subsequent siblings)
12 siblings, 0 replies; 26+ messages in thread
From: Kővágó, Zoltán @ 2015-06-01 13:23 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/wavaudio.c | 37 +++++++++++++++++++++----------------
1 file changed, 21 insertions(+), 16 deletions(-)
diff --git a/audio/wavaudio.c b/audio/wavaudio.c
index 5132aed..8acea8f 100644
--- a/audio/wavaudio.c
+++ b/audio/wavaudio.c
@@ -36,15 +36,10 @@ typedef struct WAVVoiceOut {
int total_samples;
} WAVVoiceOut;
-static struct {
+typedef struct {
struct audsettings settings;
const char *wav_path;
-} conf = {
- .settings.freq = 44100,
- .settings.nchannels = 2,
- .settings.fmt = AUD_FMT_S16,
- .wav_path = "qemu.wav"
-};
+} WAVConf;
static int wav_run_out (HWVoiceOut *hw, int live)
{
@@ -116,7 +111,8 @@ static int wav_init_out(HWVoiceOut *hw, struct audsettings *as,
0x02, 0x00, 0x44, 0xac, 0x00, 0x00, 0x10, 0xb1, 0x02, 0x00, 0x04,
0x00, 0x10, 0x00, 0x64, 0x61, 0x74, 0x61, 0x00, 0x00, 0x00, 0x00
};
- struct audsettings wav_as = conf.settings;
+ WAVConf *conf = drv_opaque;
+ struct audsettings wav_as = conf->settings;
(void) as;
(void) drv_opaque;
@@ -157,10 +153,10 @@ static int wav_init_out(HWVoiceOut *hw, struct audsettings *as,
le_store (hdr + 28, hw->info.freq << (bits16 + stereo), 4);
le_store (hdr + 32, 1 << (bits16 + stereo), 2);
- wav->f = fopen (conf.wav_path, "wb");
+ wav->f = fopen (conf->wav_path, "wb");
if (!wav->f) {
dolog ("Failed to open wave file `%s'\nReason: %s\n",
- conf.wav_path, strerror (errno));
+ conf->wav_path, strerror (errno));
g_free (wav->pcm_buf);
wav->pcm_buf = NULL;
return -1;
@@ -228,40 +224,49 @@ static int wav_ctl_out (HWVoiceOut *hw, int cmd, ...)
return 0;
}
+static WAVConf glob_conf = {
+ .settings.freq = 44100,
+ .settings.nchannels = 2,
+ .settings.fmt = AUD_FMT_S16,
+ .wav_path = "qemu.wav"
+};
+
static void *wav_audio_init (void)
{
- return &conf;
+ WAVConf *conf = g_malloc(sizeof(WAVConf));
+ *conf = glob_conf;
+ return conf;
}
static void wav_audio_fini (void *opaque)
{
- (void) opaque;
ldebug ("wav_fini");
+ g_free(opaque);
}
static struct audio_option wav_options[] = {
{
.name = "FREQUENCY",
.tag = AUD_OPT_INT,
- .valp = &conf.settings.freq,
+ .valp = &glob_conf.settings.freq,
.descr = "Frequency"
},
{
.name = "FORMAT",
.tag = AUD_OPT_FMT,
- .valp = &conf.settings.fmt,
+ .valp = &glob_conf.settings.fmt,
.descr = "Format"
},
{
.name = "DAC_FIXED_CHANNELS",
.tag = AUD_OPT_INT,
- .valp = &conf.settings.nchannels,
+ .valp = &glob_conf.settings.nchannels,
.descr = "Number of channels (1 - mono, 2 - stereo)"
},
{
.name = "PATH",
.tag = AUD_OPT_STR,
- .valp = &conf.wav_path,
+ .valp = &glob_conf.wav_path,
.descr = "Path to wave file"
},
{ /* End of list */ }
--
2.4.2
^ permalink raw reply related [flat|nested] 26+ messages in thread* [Qemu-devel] [PATCH 10/12] dsoundaudio: do not use global variables
2015-06-01 13:23 [Qemu-devel] [PATCH 00/12] Audio backend cleanup Kővágó, Zoltán
` (8 preceding siblings ...)
2015-06-01 13:23 ` [Qemu-devel] [PATCH 09/12] wavaudio: " Kővágó, Zoltán
@ 2015-06-01 13:23 ` Kővágó, Zoltán
2015-06-01 13:23 ` [Qemu-devel] [PATCH 11/12] paaudio: fix possible resource leak Kővágó, Zoltán
` (2 subsequent siblings)
12 siblings, 0 replies; 26+ messages in thread
From: Kővágó, Zoltán @ 2015-06-01 13:23 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/dsound_template.h | 18 ++++----
audio/dsoundaudio.c | 106 ++++++++++++++++++++++++++++--------------------
2 files changed, 74 insertions(+), 50 deletions(-)
diff --git a/audio/dsound_template.h b/audio/dsound_template.h
index 98276fb..85ba858 100644
--- a/audio/dsound_template.h
+++ b/audio/dsound_template.h
@@ -67,7 +67,8 @@ static int glue (dsound_lock_, TYPE) (
LPVOID *p2p,
DWORD *blen1p,
DWORD *blen2p,
- int entire
+ int entire,
+ dsound *s
)
{
HRESULT hr;
@@ -75,13 +76,14 @@ static int glue (dsound_lock_, TYPE) (
LPVOID p1 = NULL, p2 = NULL;
DWORD blen1 = 0, blen2 = 0;
DWORD flag;
+ DSoundConf *conf = &s->conf;
#ifdef DSBTYPE_IN
flag = entire ? DSCBLOCK_ENTIREBUFFER : 0;
#else
flag = entire ? DSBLOCK_ENTIREBUFFER : 0;
#endif
- for (i = 0; i < conf.lock_retries; ++i) {
+ for (i = 0; i < conf->lock_retries; ++i) {
hr = glue (IFACE, _Lock) (
buf,
pos,
@@ -96,7 +98,7 @@ static int glue (dsound_lock_, TYPE) (
if (FAILED (hr)) {
#ifndef DSBTYPE_IN
if (hr == DSERR_BUFFERLOST) {
- if (glue (dsound_restore_, TYPE) (buf)) {
+ if (glue (dsound_restore_, TYPE) (buf, s)) {
dsound_logerr (hr, "Could not lock " NAME "\n");
goto fail;
}
@@ -110,7 +112,7 @@ static int glue (dsound_lock_, TYPE) (
break;
}
- if (i == conf.lock_retries) {
+ if (i == conf->lock_retries) {
dolog ("%d attempts to lock " NAME " failed\n", i);
goto fail;
}
@@ -183,9 +185,10 @@ static int dsound_init_out(HWVoiceOut *hw, struct audsettings *as,
{
int err;
HRESULT hr;
- dsound *s = &glob_dsound;
+ dsound *s = drv_opaque;
WAVEFORMATEX wfx;
struct audsettings obt_as;
+ DSoundConf *conf = &s->conf;
#ifdef DSBTYPE_IN
const char *typ = "ADC";
DSoundVoiceIn *ds = (DSoundVoiceIn *) hw;
@@ -212,7 +215,7 @@ static int dsound_init_out(HWVoiceOut *hw, struct audsettings *as,
bd.dwSize = sizeof (bd);
bd.lpwfxFormat = &wfx;
#ifdef DSBTYPE_IN
- bd.dwBufferBytes = conf.bufsize_in;
+ bd.dwBufferBytes = conf->bufsize_in;
hr = IDirectSoundCapture_CreateCaptureBuffer (
s->dsound_capture,
&bd,
@@ -221,7 +224,7 @@ static int dsound_init_out(HWVoiceOut *hw, struct audsettings *as,
);
#else
bd.dwFlags = DSBCAPS_STICKYFOCUS | DSBCAPS_GETCURRENTPOSITION2;
- bd.dwBufferBytes = conf.bufsize_out;
+ bd.dwBufferBytes = conf->bufsize_out;
hr = IDirectSound_CreateSoundBuffer (
s->dsound,
&bd,
@@ -271,6 +274,7 @@ static int dsound_init_out(HWVoiceOut *hw, struct audsettings *as,
);
}
hw->samples = bc.dwBufferBytes >> hw->info.shift;
+ ds->s = s;
#ifdef DEBUG_DSOUND
dolog ("caps %ld, desc %ld\n",
diff --git a/audio/dsoundaudio.c b/audio/dsoundaudio.c
index e2d89fd..c8b09e2 100644
--- a/audio/dsoundaudio.c
+++ b/audio/dsoundaudio.c
@@ -41,7 +41,7 @@
/* #define DEBUG_DSOUND */
-static struct {
+typedef struct {
int lock_retries;
int restore_retries;
int getstatus_retries;
@@ -50,33 +50,22 @@ static struct {
int bufsize_out;
struct audsettings settings;
int latency_millis;
-} conf = {
- .lock_retries = 1,
- .restore_retries = 1,
- .getstatus_retries = 1,
- .set_primary = 0,
- .bufsize_in = 16384,
- .bufsize_out = 16384,
- .settings.freq = 44100,
- .settings.nchannels = 2,
- .settings.fmt = AUD_FMT_S16,
- .latency_millis = 10
-};
+} DSoundConf;
typedef struct {
LPDIRECTSOUND dsound;
LPDIRECTSOUNDCAPTURE dsound_capture;
LPDIRECTSOUNDBUFFER dsound_primary_buffer;
struct audsettings settings;
+ DSoundConf conf;
} dsound;
-static dsound glob_dsound;
-
typedef struct {
HWVoiceOut hw;
LPDIRECTSOUNDBUFFER dsound_buffer;
DWORD old_pos;
int first_time;
+ dsound *s;
#ifdef DEBUG_DSOUND
DWORD old_ppos;
DWORD played;
@@ -88,6 +77,7 @@ typedef struct {
HWVoiceIn hw;
int first_time;
LPDIRECTSOUNDCAPTUREBUFFER dsound_capture_buffer;
+ dsound *s;
} DSoundVoiceIn;
static void dsound_log_hresult (HRESULT hr)
@@ -281,12 +271,12 @@ static void print_wave_format (WAVEFORMATEX *wfx)
}
#endif
-static int dsound_restore_out (LPDIRECTSOUNDBUFFER dsb)
+static int dsound_restore_out (LPDIRECTSOUNDBUFFER dsb, dsound *s)
{
HRESULT hr;
int i;
- for (i = 0; i < conf.restore_retries; ++i) {
+ for (i = 0; i < s->conf.restore_retries; ++i) {
hr = IDirectSoundBuffer_Restore (dsb);
switch (hr) {
@@ -311,12 +301,13 @@ static int dsound_restore_out (LPDIRECTSOUNDBUFFER dsb)
#include "dsound_template.h"
#undef DSBTYPE_IN
-static int dsound_get_status_out (LPDIRECTSOUNDBUFFER dsb, DWORD *statusp)
+static int dsound_get_status_out (LPDIRECTSOUNDBUFFER dsb, DWORD *statusp,
+ dsound *s)
{
HRESULT hr;
int i;
- for (i = 0; i < conf.getstatus_retries; ++i) {
+ for (i = 0; i < s->conf.getstatus_retries; ++i) {
hr = IDirectSoundBuffer_GetStatus (dsb, statusp);
if (FAILED (hr)) {
dsound_logerr (hr, "Could not get playback buffer status\n");
@@ -324,7 +315,7 @@ static int dsound_get_status_out (LPDIRECTSOUNDBUFFER dsb, DWORD *statusp)
}
if (*statusp & DSERR_BUFFERLOST) {
- if (dsound_restore_out (dsb)) {
+ if (dsound_restore_out (dsb, s)) {
return -1;
}
continue;
@@ -376,7 +367,8 @@ static void dsound_write_sample (HWVoiceOut *hw, uint8_t *dst, int dst_len)
hw->rpos = pos % hw->samples;
}
-static void dsound_clear_sample (HWVoiceOut *hw, LPDIRECTSOUNDBUFFER dsb)
+static void dsound_clear_sample (HWVoiceOut *hw, LPDIRECTSOUNDBUFFER dsb,
+ dsound *s)
{
int err;
LPVOID p1, p2;
@@ -389,7 +381,8 @@ static void dsound_clear_sample (HWVoiceOut *hw, LPDIRECTSOUNDBUFFER dsb)
hw->samples << hw->info.shift,
&p1, &p2,
&blen1, &blen2,
- 1
+ 1,
+ s
);
if (err) {
return;
@@ -435,6 +428,7 @@ static int dsound_open (dsound *s)
WAVEFORMATEX wfx;
DSBUFFERDESC dsbd;
HWND hwnd;
+ DSoundConf *conf = &s->conf;
hwnd = GetForegroundWindow ();
hr = IDirectSound_SetCooperativeLevel (
@@ -449,11 +443,11 @@ static int dsound_open (dsound *s)
return -1;
}
- if (!conf.set_primary) {
+ if (!conf->set_primary) {
return 0;
}
- err = waveformat_from_audio_settings (&wfx, &conf.settings);
+ err = waveformat_from_audio_settings (&wfx, &conf->settings);
if (err) {
return -1;
}
@@ -514,6 +508,7 @@ static int dsound_ctl_out (HWVoiceOut *hw, int cmd, ...)
DWORD status;
DSoundVoiceOut *ds = (DSoundVoiceOut *) hw;
LPDIRECTSOUNDBUFFER dsb = ds->dsound_buffer;
+ dsound *s = ds->s;
if (!dsb) {
dolog ("Attempt to control voice without a buffer\n");
@@ -522,7 +517,7 @@ static int dsound_ctl_out (HWVoiceOut *hw, int cmd, ...)
switch (cmd) {
case VOICE_ENABLE:
- if (dsound_get_status_out (dsb, &status)) {
+ if (dsound_get_status_out (dsb, &status, s)) {
return -1;
}
@@ -531,7 +526,7 @@ static int dsound_ctl_out (HWVoiceOut *hw, int cmd, ...)
return 0;
}
- dsound_clear_sample (hw, dsb);
+ dsound_clear_sample (hw, dsb, s);
hr = IDirectSoundBuffer_Play (dsb, 0, 0, DSBPLAY_LOOPING);
if (FAILED (hr)) {
@@ -541,7 +536,7 @@ static int dsound_ctl_out (HWVoiceOut *hw, int cmd, ...)
break;
case VOICE_DISABLE:
- if (dsound_get_status_out (dsb, &status)) {
+ if (dsound_get_status_out (dsb, &status, s)) {
return -1;
}
@@ -578,6 +573,8 @@ static int dsound_run_out (HWVoiceOut *hw, int live)
DWORD wpos, ppos, old_pos;
LPVOID p1, p2;
int bufsize;
+ dsound *s = ds->s;
+ DSoundConf *conf = &s->conf;
if (!dsb) {
dolog ("Attempt to run empty with playback buffer\n");
@@ -600,14 +597,14 @@ static int dsound_run_out (HWVoiceOut *hw, int live)
len = live << hwshift;
if (ds->first_time) {
- if (conf.latency_millis) {
+ if (conf->latency_millis) {
DWORD cur_blat;
cur_blat = audio_ring_dist (wpos, ppos, bufsize);
ds->first_time = 0;
old_pos = wpos;
old_pos +=
- millis_to_bytes (&hw->info, conf.latency_millis) - cur_blat;
+ millis_to_bytes (&hw->info, conf->latency_millis) - cur_blat;
old_pos %= bufsize;
old_pos &= ~hw->info.align;
}
@@ -663,7 +660,8 @@ static int dsound_run_out (HWVoiceOut *hw, int live)
len,
&p1, &p2,
&blen1, &blen2,
- 0
+ 0,
+ s
);
if (err) {
return 0;
@@ -766,6 +764,7 @@ static int dsound_run_in (HWVoiceIn *hw)
DWORD cpos, rpos;
LPVOID p1, p2;
int hwshift;
+ dsound *s = ds->s;
if (!dscb) {
dolog ("Attempt to run without capture buffer\n");
@@ -820,7 +819,8 @@ static int dsound_run_in (HWVoiceIn *hw)
&p2,
&blen1,
&blen2,
- 0
+ 0,
+ s
);
if (err) {
return 0;
@@ -843,12 +843,26 @@ static int dsound_run_in (HWVoiceIn *hw)
return decr;
}
+static DSoundConf glob_conf = {
+ .lock_retries = 1,
+ .restore_retries = 1,
+ .getstatus_retries = 1,
+ .set_primary = 0,
+ .bufsize_in = 16384,
+ .bufsize_out = 16384,
+ .settings.freq = 44100,
+ .settings.nchannels = 2,
+ .settings.fmt = AUD_FMT_S16,
+ .latency_millis = 10
+};
+
static void dsound_audio_fini (void *opaque)
{
HRESULT hr;
dsound *s = opaque;
if (!s->dsound) {
+ g_free(s);
return;
}
@@ -859,6 +873,7 @@ static void dsound_audio_fini (void *opaque)
s->dsound = NULL;
if (!s->dsound_capture) {
+ g_free(s);
return;
}
@@ -867,17 +882,21 @@ static void dsound_audio_fini (void *opaque)
dsound_logerr (hr, "Could not release DirectSoundCapture\n");
}
s->dsound_capture = NULL;
+
+ g_free(s);
}
static void *dsound_audio_init (void)
{
int err;
HRESULT hr;
- dsound *s = &glob_dsound;
+ dsound *s = g_malloc0(sizeof(dsound));
+ s->conf = glob_conf;
hr = CoInitialize (NULL);
if (FAILED (hr)) {
dsound_logerr (hr, "Could not initialize COM\n");
+ g_free(s);
return NULL;
}
@@ -890,6 +909,7 @@ static void *dsound_audio_init (void)
);
if (FAILED (hr)) {
dsound_logerr (hr, "Could not create DirectSound instance\n");
+ g_free(s);
return NULL;
}
@@ -901,7 +921,7 @@ static void *dsound_audio_init (void)
if (FAILED (hr)) {
dsound_logerr (hr, "Could not release DirectSound\n");
}
- s->dsound = NULL;
+ g_free(s);
return NULL;
}
@@ -941,61 +961,61 @@ static struct audio_option dsound_options[] = {
{
.name = "LOCK_RETRIES",
.tag = AUD_OPT_INT,
- .valp = &conf.lock_retries,
+ .valp = &glob_conf.lock_retries,
.descr = "Number of times to attempt locking the buffer"
},
{
.name = "RESTOURE_RETRIES",
.tag = AUD_OPT_INT,
- .valp = &conf.restore_retries,
+ .valp = &glob_conf.restore_retries,
.descr = "Number of times to attempt restoring the buffer"
},
{
.name = "GETSTATUS_RETRIES",
.tag = AUD_OPT_INT,
- .valp = &conf.getstatus_retries,
+ .valp = &glob_conf.getstatus_retries,
.descr = "Number of times to attempt getting status of the buffer"
},
{
.name = "SET_PRIMARY",
.tag = AUD_OPT_BOOL,
- .valp = &conf.set_primary,
+ .valp = &glob_conf.set_primary,
.descr = "Set the parameters of primary buffer"
},
{
.name = "LATENCY_MILLIS",
.tag = AUD_OPT_INT,
- .valp = &conf.latency_millis,
+ .valp = &glob_conf.latency_millis,
.descr = "(undocumented)"
},
{
.name = "PRIMARY_FREQ",
.tag = AUD_OPT_INT,
- .valp = &conf.settings.freq,
+ .valp = &glob_conf.settings.freq,
.descr = "Primary buffer frequency"
},
{
.name = "PRIMARY_CHANNELS",
.tag = AUD_OPT_INT,
- .valp = &conf.settings.nchannels,
+ .valp = &glob_conf.settings.nchannels,
.descr = "Primary buffer number of channels (1 - mono, 2 - stereo)"
},
{
.name = "PRIMARY_FMT",
.tag = AUD_OPT_FMT,
- .valp = &conf.settings.fmt,
+ .valp = &glob_conf.settings.fmt,
.descr = "Primary buffer format"
},
{
.name = "BUFSIZE_OUT",
.tag = AUD_OPT_INT,
- .valp = &conf.bufsize_out,
+ .valp = &glob_conf.bufsize_out,
.descr = "(undocumented)"
},
{
.name = "BUFSIZE_IN",
.tag = AUD_OPT_INT,
- .valp = &conf.bufsize_in,
+ .valp = &glob_conf.bufsize_in,
.descr = "(undocumented)"
},
{ /* End of list */ }
--
2.4.2
^ permalink raw reply related [flat|nested] 26+ messages in thread* [Qemu-devel] [PATCH 11/12] paaudio: fix possible resource leak
2015-06-01 13:23 [Qemu-devel] [PATCH 00/12] Audio backend cleanup Kővágó, Zoltán
` (9 preceding siblings ...)
2015-06-01 13:23 ` [Qemu-devel] [PATCH 10/12] dsoundaudio: " Kővágó, Zoltán
@ 2015-06-01 13:23 ` Kővágó, Zoltán
2015-06-01 13:23 ` [Qemu-devel] [PATCH 12/12] coreaudio: do not use global variables where possible Kővágó, Zoltán
2015-06-01 13:38 ` [Qemu-devel] [PATCH 00/12] Audio backend cleanup Michael Tokarev
12 siblings, 0 replies; 26+ messages in thread
From: Kővágó, Zoltán @ 2015-06-01 13:23 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
qpa_audio_init did not clean up resources properly if the initialization
failed. This hopefully fixes it.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/paaudio.c | 6 +++++-
1 file changed, 5 insertions(+), 1 deletion(-)
diff --git a/audio/paaudio.c b/audio/paaudio.c
index 35e8887..fea6071 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -47,6 +47,8 @@ typedef struct {
paaudio *g;
} PAVoiceIn;
+static void qpa_audio_fini(void *opaque);
+
static void GCC_FMT_ATTR (2, 3) qpa_logerr (int err, const char *fmt, ...)
{
va_list ap;
@@ -814,6 +816,8 @@ static void *qpa_audio_init (void)
{
paaudio *g = g_malloc(sizeof(paaudio));
g->conf = glob_conf;
+ g->mainloop = NULL;
+ g->context = NULL;
g->mainloop = pa_threaded_mainloop_new ();
if (!g->mainloop) {
@@ -867,7 +871,7 @@ unlock_and_fail:
pa_threaded_mainloop_unlock (g->mainloop);
fail:
AUD_log (AUDIO_CAP, "Failed to initialize PA context");
- g_free(g);
+ qpa_audio_fini(g);
return NULL;
}
--
2.4.2
^ permalink raw reply related [flat|nested] 26+ messages in thread* [Qemu-devel] [PATCH 12/12] coreaudio: do not use global variables where possible
2015-06-01 13:23 [Qemu-devel] [PATCH 00/12] Audio backend cleanup Kővágó, Zoltán
` (10 preceding siblings ...)
2015-06-01 13:23 ` [Qemu-devel] [PATCH 11/12] paaudio: fix possible resource leak Kővágó, Zoltán
@ 2015-06-01 13:23 ` Kővágó, Zoltán
2015-06-01 13:38 ` [Qemu-devel] [PATCH 00/12] Audio backend cleanup Michael Tokarev
12 siblings, 0 replies; 26+ messages in thread
From: Kővágó, Zoltán @ 2015-06-01 13:23 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/coreaudio.c | 45 ++++++++++++++++++++++++---------------------
1 file changed, 24 insertions(+), 21 deletions(-)
diff --git a/audio/coreaudio.c b/audio/coreaudio.c
index f38fd82..6dfd63e 100644
--- a/audio/coreaudio.c
+++ b/audio/coreaudio.c
@@ -32,20 +32,16 @@
#define AUDIO_CAP "coreaudio"
#include "audio_int.h"
-struct {
+static int isAtexit;
+
+typedef struct {
int buffer_frames;
int nbuffers;
- int isAtexit;
-} conf = {
- .buffer_frames = 512,
- .nbuffers = 4,
- .isAtexit = 0
-};
+} CoreaudioConf;
typedef struct coreaudioVoiceOut {
HWVoiceOut hw;
pthread_mutex_t mutex;
- int isAtexit;
AudioDeviceID outputDeviceID;
UInt32 audioDevicePropertyBufferFrameSize;
AudioStreamBasicDescription outputStreamBasicDescription;
@@ -161,7 +157,7 @@ static inline UInt32 isPlaying (AudioDeviceID outputDeviceID)
static void coreaudio_atexit (void)
{
- conf.isAtexit = 1;
+ isAtexit = 1;
}
static int coreaudio_lock (coreaudioVoiceOut *core, const char *fn_name)
@@ -296,8 +292,7 @@ static int coreaudio_init_out(HWVoiceOut *hw, struct audsettings *as,
int err;
const char *typ = "playback";
AudioValueRange frameRange;
-
- (void) drv_opaque;
+ CoreaudioConf *conf = drv_opaque;
/* create mutex */
err = pthread_mutex_init(&core->mutex, NULL);
@@ -339,16 +334,16 @@ static int coreaudio_init_out(HWVoiceOut *hw, struct audsettings *as,
return -1;
}
- if (frameRange.mMinimum > conf.buffer_frames) {
+ if (frameRange.mMinimum > conf->buffer_frames) {
core->audioDevicePropertyBufferFrameSize = (UInt32) frameRange.mMinimum;
dolog ("warning: Upsizing Buffer Frames to %f\n", frameRange.mMinimum);
}
- else if (frameRange.mMaximum < conf.buffer_frames) {
+ else if (frameRange.mMaximum < conf->buffer_frames) {
core->audioDevicePropertyBufferFrameSize = (UInt32) frameRange.mMaximum;
dolog ("warning: Downsizing Buffer Frames to %f\n", frameRange.mMaximum);
}
else {
- core->audioDevicePropertyBufferFrameSize = conf.buffer_frames;
+ core->audioDevicePropertyBufferFrameSize = conf->buffer_frames;
}
/* set Buffer Frame Size */
@@ -382,7 +377,7 @@ static int coreaudio_init_out(HWVoiceOut *hw, struct audsettings *as,
"Could not get device buffer frame size\n");
return -1;
}
- hw->samples = conf.nbuffers * core->audioDevicePropertyBufferFrameSize;
+ hw->samples = conf->nbuffers * core->audioDevicePropertyBufferFrameSize;
/* get StreamFormat */
propertySize = sizeof(core->outputStreamBasicDescription);
@@ -446,7 +441,7 @@ static void coreaudio_fini_out (HWVoiceOut *hw)
int err;
coreaudioVoiceOut *core = (coreaudioVoiceOut *) hw;
- if (!conf.isAtexit) {
+ if (!isAtexit) {
/* stop playback */
if (isPlaying(core->outputDeviceID)) {
status = AudioDeviceStop(core->outputDeviceID, audioDeviceIOProc);
@@ -489,7 +484,7 @@ static int coreaudio_ctl_out (HWVoiceOut *hw, int cmd, ...)
case VOICE_DISABLE:
/* stop playback */
- if (!conf.isAtexit) {
+ if (!isAtexit) {
if (isPlaying(core->outputDeviceID)) {
status = AudioDeviceStop(core->outputDeviceID, audioDeviceIOProc);
if (status != kAudioHardwareNoError) {
@@ -502,28 +497,36 @@ static int coreaudio_ctl_out (HWVoiceOut *hw, int cmd, ...)
return 0;
}
+static CoreaudioConf glob_conf = {
+ .buffer_frames = 512,
+ .nbuffers = 4,
+};
+
static void *coreaudio_audio_init (void)
{
+ CoreaudioConf *conf = g_malloc(sizeof(CoreaudioConf));
+ *conf = glob_conf;
+
atexit(coreaudio_atexit);
- return &coreaudio_audio_init;
+ return conf;
}
static void coreaudio_audio_fini (void *opaque)
{
- (void) opaque;
+ g_free(opaque);
}
static struct audio_option coreaudio_options[] = {
{
.name = "BUFFER_SIZE",
.tag = AUD_OPT_INT,
- .valp = &conf.buffer_frames,
+ .valp = &glob_conf.buffer_frames,
.descr = "Size of the buffer in frames"
},
{
.name = "BUFFER_COUNT",
.tag = AUD_OPT_INT,
- .valp = &conf.nbuffers,
+ .valp = &glob_conf.nbuffers,
.descr = "Number of buffers"
},
{ /* End of list */ }
--
2.4.2
^ permalink raw reply related [flat|nested] 26+ messages in thread* Re: [Qemu-devel] [PATCH 00/12] Audio backend cleanup
2015-06-01 13:23 [Qemu-devel] [PATCH 00/12] Audio backend cleanup Kővágó, Zoltán
` (11 preceding siblings ...)
2015-06-01 13:23 ` [Qemu-devel] [PATCH 12/12] coreaudio: do not use global variables where possible Kővágó, Zoltán
@ 2015-06-01 13:38 ` Michael Tokarev
2015-06-01 13:48 ` Kővágó Zoltán
2015-06-02 8:08 ` Gerd Hoffmann
12 siblings, 2 replies; 26+ messages in thread
From: Michael Tokarev @ 2015-06-01 13:38 UTC (permalink / raw)
To: "Kővágó, Zoltán", qemu-devel; +Cc: Gerd Hoffmann
01.06.2015 16:23, Kővágó, Zoltán wrote:
> This series of patch removes the following audio backends:
> esd, fmod, sdl, winwave.
Why do you remove sdl backend?
This is my preferred backend so far, it works better
than all other backends available to me...
Thanks,
/mjt
^ permalink raw reply [flat|nested] 26+ messages in thread* Re: [Qemu-devel] [PATCH 00/12] Audio backend cleanup
2015-06-01 13:38 ` [Qemu-devel] [PATCH 00/12] Audio backend cleanup Michael Tokarev
@ 2015-06-01 13:48 ` Kővágó Zoltán
2015-06-02 8:08 ` Gerd Hoffmann
1 sibling, 0 replies; 26+ messages in thread
From: Kővágó Zoltán @ 2015-06-01 13:48 UTC (permalink / raw)
To: Michael Tokarev, qemu-devel; +Cc: Gerd Hoffmann
2015-06-01 15:38 keltezéssel, Michael Tokarev írta:
> 01.06.2015 16:23, Kővágó, Zoltán wrote:
>> This series of patch removes the following audio backends:
>> esd, fmod, sdl, winwave.
>
> Why do you remove sdl backend?
>
> This is my preferred backend so far, it works better
> than all other backends available to me...
I'm removing it because it's broken (see
http://lists.nongnu.org/archive/html/qemu-devel/2015-05/msg02198.html
for an example), and nobody had an objection about removing sdl when I
asked it on the list about a week ago
(http://lists.nongnu.org/archive/html/qemu-devel/2015-05/msg04116.html).
Thanks,
Zoltan
^ permalink raw reply [flat|nested] 26+ messages in thread
* Re: [Qemu-devel] [PATCH 00/12] Audio backend cleanup
2015-06-01 13:38 ` [Qemu-devel] [PATCH 00/12] Audio backend cleanup Michael Tokarev
2015-06-01 13:48 ` Kővágó Zoltán
@ 2015-06-02 8:08 ` Gerd Hoffmann
2015-06-02 9:12 ` Michael Tokarev
1 sibling, 1 reply; 26+ messages in thread
From: Gerd Hoffmann @ 2015-06-02 8:08 UTC (permalink / raw)
To: Michael Tokarev; +Cc: qemu-devel, "Kővágó, Zoltán"
On Mo, 2015-06-01 at 16:38 +0300, Michael Tokarev wrote:
> 01.06.2015 16:23, Kővágó, Zoltán wrote:
> > This series of patch removes the following audio backends:
> > esd, fmod, sdl, winwave.
>
> Why do you remove sdl backend?
>
> This is my preferred backend so far, it works better
> than all other backends available to me...
What other backends can you use? What are the problems?
Do you use sdl audio together with sdl ui?
cheers,
Gerd
^ permalink raw reply [flat|nested] 26+ messages in thread
* Re: [Qemu-devel] [PATCH 00/12] Audio backend cleanup
2015-06-02 8:08 ` Gerd Hoffmann
@ 2015-06-02 9:12 ` Michael Tokarev
2015-06-02 9:28 ` Gerd Hoffmann
2015-06-02 10:45 ` Peter Maydell
0 siblings, 2 replies; 26+ messages in thread
From: Michael Tokarev @ 2015-06-02 9:12 UTC (permalink / raw)
To: Gerd Hoffmann; +Cc: qemu-devel, "Kővágó, Zoltán"
02.06.2015 11:08, Gerd Hoffmann wrote:
> On Mo, 2015-06-01 at 16:38 +0300, Michael Tokarev wrote:
>> 01.06.2015 16:23, Kővágó, Zoltán wrote:
>>> This series of patch removes the following audio backends:
>>> esd, fmod, sdl, winwave.
>>
>> Why do you remove sdl backend?
>>
>> This is my preferred backend so far, it works better
>> than all other backends available to me...
>
> What other backends can you use? What are the problems?
> Do you use sdl audio together with sdl ui?
I haven't had a chance to check, planned to do that
today morning but had some other things to do. Will
check asap.
Yes, I use sdl audio with sdl ui, sdl1 version (since
sdl2 had its shortcomings in qemu, not all functionality
worked).
I also tried alsa, oss and pa, with these audio was more
distorted than with sdl. Sdl version sounds cleanly, while
alsa and pa, at least in the past, and at least if memory
serves me right, sounded like a if a good portion of noize
has been mixed in with sound, or like when sound is with
quick monmentary drops which happens at high frequency
(maybe 200Hz).
SDL audio has been the default in Debian for at least two
debian releases because of that, we explicitly used
--audio-drv-list=sdl,... to list sdl first. Now we have
pa first, but I don't really remember why we changed it.
But I have QEMU_AUDIO_DRV=sdl in my ~/.profile for a very
long time. Not that I use audio all the time, only rarely,
but I never had any issues especially with sdl (this is
ofcourse NOT to say that there are no probs in there :)
Thanks,
/mjt
^ permalink raw reply [flat|nested] 26+ messages in thread
* Re: [Qemu-devel] [PATCH 00/12] Audio backend cleanup
2015-06-02 9:12 ` Michael Tokarev
@ 2015-06-02 9:28 ` Gerd Hoffmann
2015-06-02 10:33 ` Paolo Bonzini
2015-06-02 10:45 ` Peter Maydell
1 sibling, 1 reply; 26+ messages in thread
From: Gerd Hoffmann @ 2015-06-02 9:28 UTC (permalink / raw)
To: Michael Tokarev; +Cc: qemu-devel, "Kővágó, Zoltán"
Hi,
> Yes, I use sdl audio with sdl ui, sdl1 version (since
> sdl2 had its shortcomings in qemu, not all functionality
> worked).
Does it work equally well with the gtk ui?
Possibly it makes sense to use keep sdl audio and default to it only in
case sdl ui is active (simliar to how spice audio is handled).
> I also tried alsa, oss and pa, with these audio was more
> distorted than with sdl. Sdl version sounds cleanly, while
> alsa and pa, at least in the past, and at least if memory
> serves me right, sounded like a if a good portion of noize
> has been mixed in with sound, or like when sound is with
> quick monmentary drops which happens at high frequency
> (maybe 200Hz).
Buffer overruns and underruns maybe? Possibly this is simply due to
different default values for sound buffer sizes.
> SDL audio has been the default in Debian for at least two
> debian releases because of that, we explicitly used
> --audio-drv-list=sdl,... to list sdl first. Now we have
> pa first, but I don't really remember why we changed it.
Useful data point. sdl state can't be that bad if it was default in
debian until recently ...
cheers,
Gerd
^ permalink raw reply [flat|nested] 26+ messages in thread
* Re: [Qemu-devel] [PATCH 00/12] Audio backend cleanup
2015-06-02 9:28 ` Gerd Hoffmann
@ 2015-06-02 10:33 ` Paolo Bonzini
0 siblings, 0 replies; 26+ messages in thread
From: Paolo Bonzini @ 2015-06-02 10:33 UTC (permalink / raw)
To: Gerd Hoffmann, Michael Tokarev
Cc: qemu-devel, "Kővágó, Zoltán"
On 02/06/2015 11:28, Gerd Hoffmann wrote:
> > SDL audio has been the default in Debian for at least two
> > debian releases because of that, we explicitly used
> > --audio-drv-list=sdl,... to list sdl first. Now we have
> > pa first, but I don't really remember why we changed it.
>
> Useful data point. sdl state can't be that bad if it was default in
> debian until recently ...
I guess the breakage is only for SDL 2 then.
Paolo
^ permalink raw reply [flat|nested] 26+ messages in thread
* Re: [Qemu-devel] [PATCH 00/12] Audio backend cleanup
2015-06-02 9:12 ` Michael Tokarev
2015-06-02 9:28 ` Gerd Hoffmann
@ 2015-06-02 10:45 ` Peter Maydell
2015-06-02 11:40 ` Michael Tokarev
1 sibling, 1 reply; 26+ messages in thread
From: Peter Maydell @ 2015-06-02 10:45 UTC (permalink / raw)
To: Michael Tokarev
Cc: Kővágó, Zoltán, Gerd Hoffmann, QEMU Developers
On 2 June 2015 at 10:12, Michael Tokarev <mjt@tls.msk.ru> wrote:
> I also tried alsa, oss and pa, with these audio was more
> distorted than with sdl. Sdl version sounds cleanly, while
> alsa and pa, at least in the past, and at least if memory
> serves me right, sounded like a if a good portion of noize
> has been mixed in with sound, or like when sound is with
> quick monmentary drops which happens at high frequency
> (maybe 200Hz).
>
> SDL audio has been the default in Debian for at least two
> debian releases because of that, we explicitly used
> --audio-drv-list=sdl,... to list sdl first. Now we have
> pa first, but I don't really remember why we changed it.
I think the important thing here is to avoid ever using
any of the audio subsystems' back-compatibility layers.
For instance PA can provide an alsa-compatible interface,
but if you use it you've just added an extra layer of buffering
(and a random thread) to your system. So at a distro I would
expect there to be a "this is the standard audio subsystem"
distro-wide decision, and the QEMU packages would then follow
that.
thanks
-- PMM
^ permalink raw reply [flat|nested] 26+ messages in thread
* Re: [Qemu-devel] [PATCH 00/12] Audio backend cleanup
2015-06-02 10:45 ` Peter Maydell
@ 2015-06-02 11:40 ` Michael Tokarev
2015-06-02 12:42 ` François Revol
0 siblings, 1 reply; 26+ messages in thread
From: Michael Tokarev @ 2015-06-02 11:40 UTC (permalink / raw)
To: Peter Maydell
Cc: "Kővágó, Zoltán",
Gerd Hoffmann, QEMU Developers
02.06.2015 13:45, Peter Maydell wrote:
> On 2 June 2015 at 10:12, Michael Tokarev <mjt@tls.msk.ru> wrote:
>> SDL audio has been the default in Debian for at least two
>> debian releases because of that, we explicitly used
>> --audio-drv-list=sdl,... to list sdl first. Now we have
>> pa first, but I don't really remember why we changed it.
>
> I think the important thing here is to avoid ever using
> any of the audio subsystems' back-compatibility layers.
> For instance PA can provide an alsa-compatible interface,
> but if you use it you've just added an extra layer of buffering
> (and a random thread) to your system. So at a distro I would
> expect there to be a "this is the standard audio subsystem"
> distro-wide decision, and the QEMU packages would then follow
> that.
Yes, that's true. As long as the native version actually works.
As I said before, initially it didn't, and sdl was the only
driver that worked. Maybe in 0.12, maybe even before, I don't
remember anymore, but the fact remains, in our testing only
sdl was more or less adequate, maybe due to pure luck or
phase of moon. Later native drivers started working, so we
started using native driver which is default to the OS, which
is pulseaudio (installed on all desktop environments) or,
failing that, alsa (native to linux).
Thanks,
/mjt
^ permalink raw reply [flat|nested] 26+ messages in thread
* Re: [Qemu-devel] [PATCH 00/12] Audio backend cleanup
2015-06-02 11:40 ` Michael Tokarev
@ 2015-06-02 12:42 ` François Revol
2015-06-02 15:28 ` Gerd Hoffmann
0 siblings, 1 reply; 26+ messages in thread
From: François Revol @ 2015-06-02 12:42 UTC (permalink / raw)
To: Michael Tokarev
Cc: QEMU Developers, Peter Maydell, Gerd Hoffmann,
"Kővágó, Zoltán"
On 02/06/2015 13:40, Michael Tokarev wrote:
> Yes, that's true. As long as the native version actually works.
> As I said before, initially it didn't, and sdl was the only
> driver that worked. Maybe in 0.12, maybe even before, I don't
> remember anymore, but the fact remains, in our testing only
> sdl was more or less adequate, maybe due to pure luck or
> phase of moon. Later native drivers started working, so we
> started using native driver which is default to the OS, which
> is pulseaudio (installed on all desktop environments) or,
> failing that, alsa (native to linux).
Btw, please consider that ports to new host platforms really benefit
from SDL for those things. For example we've been using SDL for the
Haiku port until last year:
http://bb.haikuports.org/haikuports/commits/df441da15a7c4a5583f6460de2ca1d570d0caee1
All other backends are either *nix only like OSS (Haiku has an OSS port
but only uses its drivers, the API is not made public), or even
Linux-only like PA or ALSA. There are more OSes out there than just
GNU/Linux :p
It's already hard enough to get stuff upstream into QEMU, having to
write native audio and graphics backends before things get usable won't
help at all.
François.
^ permalink raw reply [flat|nested] 26+ messages in thread
* Re: [Qemu-devel] [PATCH 00/12] Audio backend cleanup
2015-06-02 12:42 ` François Revol
@ 2015-06-02 15:28 ` Gerd Hoffmann
0 siblings, 0 replies; 26+ messages in thread
From: Gerd Hoffmann @ 2015-06-02 15:28 UTC (permalink / raw)
To: François Revol
Cc: Peter Maydell, Michael Tokarev, QEMU Developers,
"Kővágó, Zoltán"
On Di, 2015-06-02 at 14:42 +0200, François Revol wrote:
> On 02/06/2015 13:40, Michael Tokarev wrote:
> > Yes, that's true. As long as the native version actually works.
> > As I said before, initially it didn't, and sdl was the only
> > driver that worked. Maybe in 0.12, maybe even before, I don't
> > remember anymore, but the fact remains, in our testing only
> > sdl was more or less adequate, maybe due to pure luck or
> > phase of moon. Later native drivers started working, so we
> > started using native driver which is default to the OS, which
> > is pulseaudio (installed on all desktop environments) or,
> > failing that, alsa (native to linux).
>
>
> Btw, please consider that ports to new host platforms really benefit
> from SDL for those things. For example we've been using SDL for the
> Haiku port until last year:
>
> http://bb.haikuports.org/haikuports/commits/df441da15a7c4a5583f6460de2ca1d570d0caee1
>
> All other backends are either *nix only like OSS (Haiku has an OSS port
> but only uses its drivers, the API is not made public), or even
> Linux-only like PA or ALSA. There are more OSes out there than just
> GNU/Linux :p
>
> It's already hard enough to get stuff upstream into QEMU, having to
> write native audio and graphics backends before things get usable won't
> help at all.
Ok, makes sense.
I think we should keep sdl audio then and only drop the other three
(esd, fmod, winwave).
cheers,
Gerd
^ permalink raw reply [flat|nested] 26+ messages in thread